MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

media/base/audio_rechunker.cc

Index: media/base/audio_rechunker.cc
diff --git a/media/base/audio_rechunker.cc b/media/base/audio_rechunker.cc
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index 0000000000000000000000000000000000000000..72326028db69535f299b1dc16c0c55aefe3b585d
--- /dev/null
+++ b/media/base/audio_rechunker.cc
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+// Copyright 2016 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.
+
+#include "media/base/audio_rechunker.h"
+
+namespace media {
+
+AudioRechunker::AudioRechunker(base::TimeDelta output_duration,
+                               const RechunkedAudioCallback& callback)
+    : output_duration_(output_duration), callback_(callback),
+      sample_rate_(0), output_frames_(0) {
+  DCHECK_GT(output_duration_, base::TimeDelta());
+  DCHECK(!callback_.is_null());
+}
+
+AudioRechunker::~AudioRechunker() {}
+
+bool AudioRechunker::SetSampleRate(int sample_rate) {
+  DCHECK_GT(sample_rate, 0);
+
+  sample_rate_ = sample_rate;
+  audio_queue_.reset();
+  pending_frames_= 0;
+
+  const int64_t numerator = sample_rate * output_duration_.InMicroseconds();
+  output_frames_ = numerator / base::Time::kMicrosecondsPerSecond;
+  DCHECK_GT(output_frames_, 0);
+  return ((numerator % base::Time::kMicrosecondsPerSecond) == 0);
+}
+
+void AudioRechunker::Push(const AudioBus& input_bus,
+                          base::TimeDelta reference_timestamp) {
+  DCHECK_GT(output_frames_, 0);
+
+  // Fast path: No buffering required.
+  if ((pending_frames_ == 0) && (input_bus.frames() == output_frames_)) {
+    callback_.Run(input_bus, reference_timestamp);
+    return;
+  }
+
+  // Lazy-create the |audio_queue_| if needed.
+  if (!audio_queue_ || audio_queue_->channels() != input_bus.channels())
+    audio_queue_ = AudioBus::Create(input_bus.channels(), output_frames_);
+
+  // Adjust |reference_timestamp| to be that of the first sample in
+  // |audio_queue_| since that will be the first sample delivered via the
+  // |callback_|.
+  reference_timestamp -= base::TimeDelta::FromMicroseconds(
+      pending_frames_ * base::Time::kMicrosecondsPerSecond / sample_rate_);
+
+  // Repeatedly fill up |audio_queue_| with more sample frames from |input_bus|
+  // and deliver batches until all sample frames in |input_bus| have been
+  // consumed.
+  int source_pos = 0;
+  do {
+    // Attempt to fill |audio_queue_| completely.
+    const int frames_to_push =
+        std::min(static_cast<int>(input_bus.frames() - source_pos),
+                 output_frames_ - pending_frames_);
+    if (frames_to_push > 0) {
+      DVLOG(2) << "Enqueuing " << frames_to_push << " frames.";
+      input_bus.CopyPartialFramesTo(source_pos, frames_to_push, pending_frames_,
+                                    audio_queue_.get());
+      pending_frames_ += frames_to_push;
+      source_pos += frames_to_push;
+    }
+
+    // If |audio_queue_| has been filled completely, deliver the re-chunked
+    // audio to the consumer.
+    if (pending_frames_ == output_frames_) {
+      DVLOG(2) << "Delivering another " << pending_frames_ << " frames.";
+      callback_.Run(*audio_queue_, reference_timestamp);
+      reference_timestamp += output_duration_;
+      pending_frames_ = 0;
+    }
+  } while (source_pos < input_bus.frames());
+}
+
+}  // namespace media