Add an initial revision of an audio server.

services/media/audio/audio_track_impl.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 <algorithm>
+#include <limits>
+
+#include "base/logging.h"
+#include "mojo/services/media/common/cpp/linear_transform.h"
+#include "services/media/audio/audio_output_manager.h"
+#include "services/media/audio/audio_server_impl.h"
+#include "services/media/audio/audio_track_impl.h"
+#include "services/media/audio/audio_track_to_output_link.h"
+
+namespace mojo {
+namespace media {
+namespace audio {
+
+constexpr size_t AudioTrackImpl::PTS_FRACTIONAL_BITS;
+
+// TODO(johngro): If there is ever a better way to do this type of static-table
+// initialization using mojom generated structs, we should switch to it.
+static const struct {
+  LpcmSampleFormat sample_format;
+  uint8_t min_samples_per_frame;
+  uint8_t max_samples_per_frame;
+  uint32_t min_frames_per_second;
+  uint32_t max_frames_per_second;
+} kSupportedLpcmTypeSets[] = {
+  {
+    .sample_format = LpcmSampleFormat::UNSIGNED_8,
+    .min_samples_per_frame = 1,
+    .max_samples_per_frame = 2,
+    .min_frames_per_second = 1000,
+    .max_frames_per_second = 48000,
+  },
+  {
+    .sample_format = LpcmSampleFormat::SIGNED_16,
+    .min_samples_per_frame = 1,
+    .max_samples_per_frame = 2,
+    .min_frames_per_second = 1000,
+    .max_frames_per_second = 48000,
+  },
+};
+
+AudioTrackImpl::AudioTrackImpl(InterfaceRequest<AudioTrack> iface,
+                             AudioServerImpl* owner)
+  : owner_(owner),
+    binding_(this),
+    pipe_(this, owner) {
+  CHECK(nullptr != owner_);
+  binding_.Bind(iface.Pass());
+}
+
+AudioTrackImpl::~AudioTrackImpl() {
+}
+
+AudioTrackImplPtr AudioTrackImpl::Create(InterfaceRequest<AudioTrack> iface,
+                                       AudioServerImpl* owner) {
+  AudioTrackImplPtr ret(new AudioTrackImpl(iface.Pass(), owner));
+  ret->weak_this_ = ret;
+  return ret;
+}
+
+void AudioTrackImpl::Describe(const DescribeCallback& cbk) {
+  // Build a minimal descriptor
+  //
+  // TODO(johngro): one day, we need to make this description much more rich and
+  // fully describe our capabilities, based on things like what outputs are
+  // available, the class of hardware we are on, and what options we were
+  // compiled with.
+  //
+  // For now, it would be nice to just be able to have a static const tree of
+  // capabilities in this translational unit which we could use to construct our
+  // message, but the nature of the structures generated by the C++ bindings
+  // make this difficult.  For now, we just create a trivial descriptor entierly
+  // by hand.
+  AudioTrackDescriptorPtr desc(AudioTrackDescriptor::New());
+
+  desc->supported_media_types =
+    Array<MediaTypeSetPtr>::New(arraysize(kSupportedLpcmTypeSets));
+
+  for (size_t i = 0; i < desc->supported_media_types.size(); ++i) {
+    const MediaTypeSetPtr& mts =
+      (desc->supported_media_types[i] = MediaTypeSet::New());
+
+    mts->scheme  = MediaTypeScheme::LPCM;
+    mts->details = MediaTypeSetDetails::New();
+
+    const auto& s = kSupportedLpcmTypeSets[i];
+    LpcmMediaTypeSetDetailsPtr lpcm_detail = LpcmMediaTypeSetDetails::New();
+
+    lpcm_detail->sample_format         = s.sample_format;
+    lpcm_detail->min_samples_per_frame = s.min_samples_per_frame;
+    lpcm_detail->max_samples_per_frame = s.max_samples_per_frame;
+    lpcm_detail->min_frames_per_second = s.min_frames_per_second;
+    lpcm_detail->max_frames_per_second = s.max_frames_per_second;
+    mts->details->set_lpcm(lpcm_detail.Pass());
+  }
+
+  cbk.Run(desc.Pass());
+}
+
+void AudioTrackImpl::Configure(AudioTrackConfigurationPtr configuration,
+                              InterfaceRequest<MediaPipe> req,
+                              const ConfigureCallback&    cbk) {
+  // Are we already configured?
+  if (pipe_.IsInitialized()) {
+    cbk.Run(MediaResult::BAD_STATE);
+    return;
+  }
+
+  // Check the requested configuration.
+  if ((configuration->media_type->scheme != MediaTypeScheme::LPCM) ||
+      (!configuration->media_type->details->is_lpcm())) {
+    cbk.Run(MediaResult::UNSUPPORTED_CONFIG);
+    return;
+  }
+
+  // Search our supported configuration sets to find one compatible with this
+  // request.
+  auto& cfg = configuration->media_type->details->get_lpcm();
+  size_t i;
+  for (i = 0; i < arraysize(kSupportedLpcmTypeSets); ++i) {
+    const auto& cfg_set = kSupportedLpcmTypeSets[i];
+
+    if ((cfg->sample_format == cfg_set.sample_format) &&
+        (cfg->samples_per_frame >= cfg_set.min_samples_per_frame) &&
+        (cfg->samples_per_frame <= cfg_set.max_samples_per_frame) &&
+        (cfg->frames_per_second >= cfg_set.min_frames_per_second) &&
+        (cfg->frames_per_second <= cfg_set.max_frames_per_second)) {
+      break;
+    }
+  }
+
+  if (i >= arraysize(kSupportedLpcmTypeSets)) {
+    cbk.Run(MediaResult::UNSUPPORTED_CONFIG);
+    return;
+  }
+
+  // Sanity check the ratio which relates audio frames to media time.
+  int32_t  numerator   = static_cast<int32_t>(configuration->audio_frame_ratio);
+  uint32_t denominator = static_cast<int32_t>(configuration->media_time_ratio);
+  if ((numerator < 1) || (denominator < 1)) {
+    cbk.Run(MediaResult::INVALID_ARGUMENT);
+    return;
+  }
+
+
+  // Figure out the rate we need to scale by in order to produce our fixed
+  // point timestamps.
+  LinearTransform::Ratio frac_scale(1 << PTS_FRACTIONAL_BITS, 1);
+  LinearTransform::Ratio frame_scale(LinearTransform::Ratio(numerator,
+                                                            denominator));
+  bool no_loss = LinearTransform::Ratio::Compose(frac_scale,
+                                                 frame_scale,
+                                                 &frame_to_media_ratio_);
+  if (!no_loss) {
+    cbk.Run(MediaResult::INVALID_ARGUMENT);
+    return;
+  }
+
+  // Figure out how many bytes we need to hold the requested number of nSec of
+  // audio.
+  switch (cfg->sample_format) {
+    case LpcmSampleFormat::UNSIGNED_8:
+      bytes_per_frame_ = 1;
+      break;
+
+    case LpcmSampleFormat::SIGNED_16:
+      bytes_per_frame_ = 2;
+      break;
+
+    case LpcmSampleFormat::SIGNED_24_IN_32:
+      bytes_per_frame_ = 4;
+      break;
+
+    default:
+      DCHECK(false);
+      bytes_per_frame_ = 2;
+      break;
+  }
+  bytes_per_frame_ *= cfg->samples_per_frame;
+
+  // Overflow trying to convert from frames to bytes?
+  uint64_t requested_frames = configuration->max_frames;
+  if (requested_frames >
+     (std::numeric_limits<size_t>::max() / bytes_per_frame_)) {
+    cbk.Run(MediaResult::INSUFFICIENT_RESOURCES);
+    return;
+  }
+
+  size_t requested_bytes = (requested_frames * bytes_per_frame_);
+
+  // Attempt to initialize our shared buffer and bind it to our interface
+  // request.
+  if (pipe_.Init(req.Pass(), requested_bytes) != MOJO_RESULT_OK) {
+    cbk.Run(MediaResult::INSUFFICIENT_RESOURCES);
+    return;
+  }
+
+  // Stash our configuration.
+  format_ = cfg.Pass();
+
+  // Have the audio output manager initialize our set of outputs.  Note; there
+  // is currently no need for a lock here.  Methods called from our user-facing
+  // interfaces are seriailzed by nature of the mojo framework, and none of the
+  // output manager's threads should ever need to manipulate the set.  Cleanup
+  // of outputs which have gone away is currently handled in a lazy fashion when
+  // the track fails to promote its weak reference during an operation involving
+  // its outputs.
+  //
+  // TODO(johngro): someday, we will need to deal with recalculating properties
+  // which depend on a track's current set of outputs (for example, the minimum
+  // latency).  This will probably be done using a dirty flag in the track
+  // implementations, and scheduling a job to recalculate the properties for the
+  // dirty tracks and notify the users as appropriate.
+
+  // If we cannot promote our own weak pointer, something is seriously wrong.
+  AudioTrackImplPtr strong_this(weak_this_.lock());
+  DCHECK(strong_this);
+  DCHECK(owner_);
+  owner_->GetOutputManager().SelectOutputsForTrack(strong_this);
+
+  // Done
+  cbk.Run(MediaResult::OK);
+}
+
+void AudioTrackImpl::GetRateControl(InterfaceRequest<RateControl> req,
+                                   const GetRateControlCallback& cbk) {
+  cbk.Run(rate_control_.Bind(req.Pass()));
+}
+
+void AudioTrackImpl::AddOutput(AudioTrackToOutputLinkPtr link) {
+  // TODO(johngro): assert that we are on the main message loop thread.
+  DCHECK(link);
+  auto res = outputs_.emplace(link);
+  DCHECK(res.second);
+}
+
+void AudioTrackImpl::RemoveOutput(AudioTrackToOutputLinkPtr link) {
+  // TODO(johngro): assert that we are on the main message loop thread.
+  DCHECK(link);
+
+  auto iter = outputs_.find(link);
+  if (iter != outputs_.end()) {
+    outputs_.erase(iter);
+  } else {
+    // TODO(johngro): that's odd.  I can't think of a reason why we we should
+    // not be able to find this link in our set of outputs... should we log
+    // something about this?
+    DCHECK(false);
+  }
+}
+
+void AudioTrackImpl::OnPacketReceived(AudioPipe::AudioPacketRefPtr packet) {
+  DCHECK(packet);
+  for (const auto& output : outputs_) {
+    DCHECK(output);
+    output->PushToPendingQueue(packet);
+  }
+}
+
+void AudioTrackImpl::OnFlushRequested(const MediaPipe::FlushCallback& cbk) {
+  for (const auto& output : outputs_) {
+    DCHECK(output);
+    output->FlushPendingQueue();
+  }
+  cbk.Run(MediaResult::OK);
+}
+
+}  // namespace audio
+}  // namespace media
+}  // namespace mojo