From 23730a6e56a168d1879203e4b3819bb36e3d8f1f Mon Sep 17 00:00:00 2001
From: "Torne (Richard Coles)" <torne@google.com>
Date: Fri, 21 Mar 2014 14:25:57 +0000
Subject: Merge from Chromium at DEPS revision 258528

This commit was generated by merge_to_master.py.

Change-Id: Id197cbdde2c2881915f096c0426c741416884783
---
 media/base/audio_converter.cc | 49 +++++++++++++++++++++++--------------------
 1 file changed, 26 insertions(+), 23 deletions(-)

(limited to 'media/base/audio_converter.cc')

diff --git a/media/base/audio_converter.cc b/media/base/audio_converter.cc
index d0c45136da..cdc6642b5f 100644
--- a/media/base/audio_converter.cc
+++ b/media/base/audio_converter.cc
@@ -25,7 +25,8 @@ namespace media {
 AudioConverter::AudioConverter(const AudioParameters& input_params,
                                const AudioParameters& output_params,
                                bool disable_fifo)
-    : downmix_early_(false),
+    : chunk_size_(output_params.frames_per_buffer()),
+      downmix_early_(false),
       resampler_frame_delay_(0),
       input_channel_count_(input_params.channels()) {
   CHECK(input_params.IsValid());
@@ -41,16 +42,6 @@ AudioConverter::AudioConverter(const AudioParameters& input_params,
 
     // Pare off data as early as we can for efficiency.
     downmix_early_ = input_params.channels() > output_params.channels();
-    if (downmix_early_) {
-      DVLOG(1) << "Remixing channel layout prior to resampling.";
-      // |unmixed_audio_| will be allocated on the fly.
-    } else {
-      // Instead, if we're not downmixing early we need a temporary AudioBus
-      // which matches the input channel count but uses the output frame size
-      // since we'll mix into the AudioBus from the output stream.
-      unmixed_audio_ = AudioBus::Create(
-          input_params.channels(), output_params.frames_per_buffer());
-    }
   }
 
   // Only resample if necessary since it's expensive.
@@ -86,12 +77,11 @@ AudioConverter::AudioConverter(const AudioParameters& input_params,
   if (input_params.frames_per_buffer() != output_params.frames_per_buffer()) {
     DVLOG(1) << "Rebuffering from " << input_params.frames_per_buffer()
              << " to " << output_params.frames_per_buffer();
+    chunk_size_ = input_params.frames_per_buffer();
     audio_fifo_.reset(new AudioPullFifo(
-        downmix_early_ ? output_params.channels() :
-            input_params.channels(),
-        input_params.frames_per_buffer(), base::Bind(
-            &AudioConverter::SourceCallback,
-            base::Unretained(this))));
+        downmix_early_ ? output_params.channels() : input_params.channels(),
+        chunk_size_,
+        base::Bind(&AudioConverter::SourceCallback, base::Unretained(this))));
   }
 }
 
@@ -119,6 +109,12 @@ void AudioConverter::Reset() {
     resampler_->Flush();
 }
 
+int AudioConverter::ChunkSize() const {
+  if (!resampler_)
+    return chunk_size_;
+  return resampler_->ChunkSize();
+}
+
 void AudioConverter::ConvertWithDelay(const base::TimeDelta& initial_delay,
                                       AudioBus* dest) {
   initial_delay_ = initial_delay;
@@ -133,6 +129,10 @@ void AudioConverter::ConvertWithDelay(const base::TimeDelta& initial_delay,
   // resampling we can save a lot of processing time.  Vice versa, we don't want
   // to increase the channel count prior to resampling for the same reason.
   bool needs_mixing = channel_mixer_ && !downmix_early_;
+
+  if (needs_mixing)
+    CreateUnmixedAudioIfNecessary(dest->frames());
+
   AudioBus* temp_dest = needs_mixing ? unmixed_audio_.get() : dest;
   DCHECK(temp_dest);
 
@@ -168,13 +168,11 @@ void AudioConverter::SourceCallback(int fifo_frame_delay, AudioBus* dest) {
         AudioBus::Create(input_channel_count_, dest->frames());
   }
 
-  if (needs_downmix &&
-      (!unmixed_audio_ || unmixed_audio_->frames() != dest->frames())) {
-    // If we're downmixing early we need a temporary AudioBus which matches
-    // the the input channel count and input frame size since we're passing
-    // |unmixed_audio_| directly to the |source_callback_|.
-    unmixed_audio_ = AudioBus::Create(input_channel_count_, dest->frames());
-  }
+  // If we're downmixing early we need a temporary AudioBus which matches
+  // the the input channel count and input frame size since we're passing
+  // |unmixed_audio_| directly to the |source_callback_|.
+  if (needs_downmix)
+    CreateUnmixedAudioIfNecessary(dest->frames());
 
   AudioBus* temp_dest = needs_downmix ? unmixed_audio_.get() : dest;
 
@@ -243,4 +241,9 @@ void AudioConverter::ProvideInput(int resampler_frame_delay, AudioBus* dest) {
     SourceCallback(0, dest);
 }
 
+void AudioConverter::CreateUnmixedAudioIfNecessary(int frames) {
+  if (!unmixed_audio_ || unmixed_audio_->frames() != frames)
+    unmixed_audio_ = AudioBus::Create(input_channel_count_, frames);
+}
+
 }  // namespace media
-- 
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