Switch to pcm_low_latency and enable resampling.

media/base/audio_renderer_mixer.cc

Index: media/base/audio_renderer_mixer.cc
diff --git a/media/base/audio_renderer_mixer.cc b/media/base/audio_renderer_mixer.cc
index 5b89aa41daebe06c3e5fcd40b1105db64a6f3fb8..56524cac2ab4ecd812ab6ab259236785b48813dd 100644
--- a/media/base/audio_renderer_mixer.cc
+++ b/media/base/audio_renderer_mixer.cc
@@ -5,16 +5,39 @@
 #include "media/base/audio_renderer_mixer.h"
 
 #include "base/logging.h"
+#include "media/audio/audio_util.h"
 
 namespace media {
 
 AudioRendererMixer::AudioRendererMixer(
     const AudioParameters& params, const scoped_refptr<AudioRendererSink>& sink)
-    : audio_parameters_(params),
-      audio_sink_(sink) {
-  // TODO(dalecurtis): Once we have resampling we'll need to pass on a different
-  // set of AudioParameters than the ones we're given.
-  audio_sink_->Initialize(audio_parameters_, this);
+    : audio_sink_(sink),
+      current_audio_delay_milliseconds_(0) {
+  // Create output parameters for passing on to the output sink.
+  // TODO(dalecurtis): There's a lot of buffer trickery to figure out here...
+  // TODO(dalecurtis): What about the values set in AudioRendererImpl?
+  // TODO(dalecurtis): All this buffer logic should be shared between WebRTC,
+  // Pepper, etc.
+  int output_sample_rate = GetAudioHardwareSampleRate();
+  output_parameters_ = AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, params.channel_layout(),
+      output_sample_rate, 16, output_sample_rate / 100);

[Chris Rogers, 2012/07/02 22:53:50]

Instead of output_sample_rate / 100, better to use:

GetAudioHardwareBufferSize()

[DaleCurtis, 2012/07/12 00:40:44]

Done.

+
+  // Only resample if necessary since it's expensive.
+  if (params.sample_rate() != output_sample_rate)

[Chris Rogers, 2012/07/02 22:53:50]

Maybe some basic sanity checking on params.sample_rate() to protect against wild
values (here and/or internal to MultiChannelResampler)

I thought we had pre-defined constants for min/max

[DaleCurtis, 2012/07/12 00:40:44]

Done.

+    resampler_.reset(new MultiChannelResampler(
+        this, params.sample_rate() / static_cast<float>(output_sample_rate),
+        output_parameters_.channels()));
+
+  // Preallocate staging area for collecting each mixer input's audio data.
+  // TODO(dalecurtis): If we switch to AVX/SSE optimization, we'll need to
+  // allocate these on 32-byte boundaries and ensure they're sized % 32 bytes.
+  mixer_input_audio_data_.reserve(output_parameters_.channels());
+  for (int i = 0; i < output_parameters_.channels(); ++i)
+    mixer_input_audio_data_.push_back(
+        new float[output_parameters_.frames_per_buffer()]);
+
+  audio_sink_->Initialize(output_parameters_, this);
   audio_sink_->Start();
 }
 
@@ -22,6 +45,11 @@ AudioRendererMixer::~AudioRendererMixer() {
   // AudioRendererSinks must be stopped before being destructed.
   audio_sink_->Stop();
 
+  // Clean up |mixer_input_audio_data_|.
+  for (size_t i = 0; i < mixer_input_audio_data_.size(); ++i)
+    delete [] mixer_input_audio_data_[i];
+  mixer_input_audio_data_.clear();
+
   // Ensures that all mixer inputs have stopped themselves prior to destruction
   // and have called RemoveMixerInput().
   DCHECK_EQ(mixer_inputs_.size(), 0U);
@@ -42,11 +70,29 @@ void AudioRendererMixer::RemoveMixerInput(
 int AudioRendererMixer::Render(const std::vector<float*>& audio_data,
                                int number_of_frames,
                                int audio_delay_milliseconds) {
+  current_audio_delay_milliseconds_ = audio_delay_milliseconds;
+
+  if (resampler_ != NULL)
+    resampler_->Resample(audio_data, number_of_frames);
+  else
+    ProvideInput(audio_data, number_of_frames);
+
+  // Always return the full number of frames requested, we padded with silence
+  // if we couldn't get enough data.

[Chris Rogers, 2012/07/02 22:53:50]

nit: fix up grammar here

[DaleCurtis, 2012/07/12 00:40:44]

Done.

+  return number_of_frames;
+}
+
+void AudioRendererMixer::ProvideInput(const std::vector<float*>& audio_data,
+                                      int number_of_frames) {
   base::AutoLock auto_lock(mixer_inputs_lock_);
 
+  // Sanity check our inputs.
+  DCHECK_LE(number_of_frames, output_parameters_.frames_per_buffer());
+  DCHECK_EQ(static_cast<int>(audio_data.size()), output_parameters_.channels());
+
   // Zero |audio_data| so we're mixing into a clean buffer and return silence if
   // we couldn't get enough data from our inputs.
-  for (int i = 0; i < audio_parameters_.channels(); ++i)
+  for (size_t i = 0; i < audio_data.size(); ++i)
     memset(audio_data[i], 0, number_of_frames * sizeof(*audio_data[i]));
 
   // Have each mixer render its data into an output buffer then mix the result.
@@ -57,35 +103,29 @@ int AudioRendererMixer::Render(const std::vector<float*>& audio_data,
     double volume;
     input->GetVolume(&volume);
 
-    // Nothing to do if the input isn't playing or the volume is zero.
-    if (!input->playing() || volume == 0.0f)
+    // Nothing to do if the input isn't playing.
+    if (!input->playing())
       continue;
 
-    const std::vector<float*>& mixer_input_audio_data = input->audio_data();
-
     int frames_filled = input->callback()->Render(

[Chris Rogers, 2012/07/02 22:53:50]

small nit: this isn't new to this patch, but it still strikes me as odd that we
have a callback() method on the input.  Why don't we just implement a
ARMI::ProvideInput() method which internally accesses the callback.  Up to you,
but this seems a little round-about

-        mixer_input_audio_data, number_of_frames, audio_delay_milliseconds);
+        mixer_input_audio_data_, number_of_frames,
+        current_audio_delay_milliseconds_);
     if (frames_filled == 0)
       continue;
 
-    // TODO(dalecurtis): Resample audio data.
-
     // Volume adjust and mix each mixer input into |audio_data| after rendering.
     // TODO(dalecurtis): Optimize with NEON/SSE/AVX vector_fmac from FFmpeg.
-    for (int j = 0; j < audio_parameters_.channels(); ++j) {
+    for (size_t j = 0; j < audio_data.size(); ++j) {
       float* dest = audio_data[j];
-      float* source = mixer_input_audio_data[j];
+      float* source = mixer_input_audio_data_[j];
       for (int k = 0; k < frames_filled; ++k)
         dest[k] += source[k] * static_cast<float>(volume);
     }
 
     // No need to clamp values as InterleaveFloatToInt() will take care of this
     // for us later when data is transferred to the browser process.
+    // TODO(dalecurtis): Does the resampler need values betwen [-1, 1] ?

[Chris Rogers, 2012/07/02 22:53:50]

The resampler will be fine.  I'd just entirely remove the comments on 125:127

   }
-
-  // Always return the full number of frames requested, padded with silence if
-  // we couldn't get enough data.
-  return number_of_frames;
 }
 
 void AudioRendererMixer::OnRenderError() {