Add an initial revision of an audio server.

services/media/audio/audio_pipe.cc

+// Copyright 2015 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 <limits>
+#include <vector>
+
+#include "services/media/audio/audio_pipe.h"
+#include "services/media/audio/audio_server_impl.h"
+#include "services/media/audio/audio_track_impl.h"
+
+namespace mojo {
+namespace media {
+namespace audio {
+
+AudioPipe::AudioPacketRef::AudioPacketRef(
+    MediaPacketStatePtr state,
+    AudioServerImpl* server,
+    std::vector<Region>&& regions,  // NOLINT(build/c++11)
+    int64_t start_pts,
+    int64_t end_pts)
+  : state_(std::move(state)),
+    server_(server),
+    regions_(std::move(regions)),
+    start_pts_(start_pts),
+    end_pts_(end_pts)  {
+  DCHECK(state_);
+  DCHECK(server_);
+}
+
+AudioPipe::AudioPacketRef::~AudioPacketRef() {
+  DCHECK(server_);
+  server_->SchedulePacketCleanup(std::move(state_));
+}
+
+AudioPipe::AudioPipe(AudioTrackImpl* owner,
+                     AudioServerImpl* server)
+  : owner_(owner),
+    server_(server) {
+  DCHECK(owner_);
+  DCHECK(server_);
+}
+
+AudioPipe::~AudioPipe() {}
+
+void AudioPipe::OnPacketReceived(MediaPacketStatePtr state) {
+  const MediaPacketPtr& packet = state->GetPacket();
+  DCHECK(packet);
+  DCHECK(packet->payload);
+  DCHECK(owner_);
+
+  // Start by making sure that we are regions we are receiving are made from an
+  // integral number of audio frames.  Count the total number of frames in the
+  // process.
+  //
+  // TODO(johngro): Someday, automatically enforce this using
+  // alignment/allocation restrictions at the MediaPipe level of things.
+  uint32_t frame_count = 0;
+  uint32_t frame_size = owner_->BytesPerFrame();
+  const MediaPacketRegionPtr* region = &packet->payload;
+  size_t ndx = 0;
+  std::vector<AudioPacketRef::Region> regions;
+
+  regions.reserve(packet->extra_payload.size() + 1);
+
+  DCHECK(frame_size);
+  while (true) {
+    if ((frame_size > 1) && ((*region)->length % frame_size)) {
+      state->SetResult(MediaResult::INVALID_ARGUMENT);
+      return;
+    }
+
+    frame_count += ((*region)->length / frame_size);
+    if (frame_count > (std::numeric_limits<uint32_t>::max() >>
+                       AudioTrackImpl::PTS_FRACTIONAL_BITS)) {
+      state->SetResult(MediaResult::INVALID_ARGUMENT);
+      return;
+    }
+
+    regions.emplace_back(
+        static_cast<const uint8_t*>(buffer()) + (*region)->offset,
+        frame_count << AudioTrackImpl::PTS_FRACTIONAL_BITS);
+
+    if (ndx >= packet->extra_payload.size()) {
+      break;
+    }
+
+    region = &(packet->extra_payload[ndx]);
+    ndx++;
+  }
+
+  // Figure out the starting PTS.
+  int64_t start_pts;
+  if (packet->pts != MediaPacket::kNoTimestamp) {
+    // The user provided an explicit PTS for this audio.  Transform it into
+    // units of fractional frames.
+    LinearTransform tmp(0, owner_->FractionalFrameToMediaTimeRatio(), 0);
+    if (!tmp.DoForwardTransform(packet->pts, &start_pts)) {
+      state->SetResult(MediaResult::INTERNAL_ERROR);
+      return;
+    }
+  } else {
+    // No PTS was provided.  Use the end time of the last audio packet, if
+    // known.  Otherwise, just assume a media time of 0.
+    start_pts = next_pts_known_ ? next_pts_ : 0;
+  }
+
+  // The end pts is the value we will use for the next packet's start PTS, if
+  // the user does not provide an explicit PTS.
+  int64_t pts_delta = (static_cast<int64_t>(frame_count)
+                    << AudioTrackImpl::PTS_FRACTIONAL_BITS);
+  next_pts_ = start_pts + pts_delta;
+  next_pts_known_ = true;
+
+  owner_->OnPacketReceived(AudioPacketRefPtr(
+        new AudioPacketRef(std::move(state),
+                           server_,
+                           std::move(regions),
+                           start_pts,
+                           next_pts_)));
+}
+
+void AudioPipe::OnFlushRequested(const FlushCallback& cbk) {
+  DCHECK(owner_);
+  owner_->OnFlushRequested(cbk);
+  next_pts_known_ = false;
+}
+
+}  // namespace audio
+}  // namespace media
+}  // namespace mojo