make pulseaudio available for all posix platforms because it's not linux only

media/audio/linux/pulse_output.cc

Index: media/audio/linux/pulse_output.cc
diff --git a/media/audio/linux/pulse_output.cc b/media/audio/linux/pulse_output.cc
deleted file mode 100644
index ddd23ca7a8cc3818d43f4776fe23f459011f3a42..0000000000000000000000000000000000000000
--- a/media/audio/linux/pulse_output.cc
+++ /dev/null
@@ -1,420 +0,0 @@
-// Copyright (c) 2011 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/audio/linux/pulse_output.h"
-
-#include "base/bind.h"
-#include "base/message_loop.h"
-#include "media/audio/audio_parameters.h"
-#include "media/audio/audio_util.h"
-#include "media/audio/linux/audio_manager_linux.h"
-#include "media/base/data_buffer.h"
-#include "media/base/seekable_buffer.h"
-
-static pa_sample_format_t BitsToPASampleFormat(int bits_per_sample) {
-  switch (bits_per_sample) {
-    // Unsupported sample formats shown for reference.  I am assuming we want
-    // signed and little endian because that is what we gave to ALSA.
-    case 8:
-      return PA_SAMPLE_U8;
-      // Also 8-bits: PA_SAMPLE_ALAW and PA_SAMPLE_ULAW
-    case 16:
-      return PA_SAMPLE_S16LE;
-      // Also 16-bits: PA_SAMPLE_S16BE (big endian).
-    case 24:
-      return PA_SAMPLE_S24LE;
-      // Also 24-bits: PA_SAMPLE_S24BE (big endian).
-      // Other cases: PA_SAMPLE_24_32LE (in LSB of 32-bit field, little endian),
-      //          and PA_SAMPLE_24_32BE (in LSB of 32-bit field, big endian),
-    case 32:
-      return PA_SAMPLE_S32LE;
-      // Also 32-bits: PA_SAMPLE_S32BE (big endian),
-      //               PA_SAMPLE_FLOAT32LE (floating point little endian),
-      //           and PA_SAMPLE_FLOAT32BE (floating point big endian).
-    default:
-      return PA_SAMPLE_INVALID;
-  }
-}
-
-static pa_channel_position ChromiumToPAChannelPosition(Channels channel) {
-  switch (channel) {
-    // PulseAudio does not differentiate between left/right and
-    // stereo-left/stereo-right, both translate to front-left/front-right.
-    case LEFT:
-    case STEREO_LEFT:
-      return PA_CHANNEL_POSITION_FRONT_LEFT;
-    case RIGHT:
-    case STEREO_RIGHT:
-      return PA_CHANNEL_POSITION_FRONT_RIGHT;
-    case CENTER:
-      return PA_CHANNEL_POSITION_FRONT_CENTER;
-    case LFE:
-      return PA_CHANNEL_POSITION_LFE;
-    case BACK_LEFT:
-      return PA_CHANNEL_POSITION_REAR_LEFT;
-    case BACK_RIGHT:
-      return PA_CHANNEL_POSITION_REAR_RIGHT;
-    case LEFT_OF_CENTER:
-      return PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
-    case RIGHT_OF_CENTER:
-      return PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
-    case BACK_CENTER:
-      return PA_CHANNEL_POSITION_REAR_CENTER;
-    case SIDE_LEFT:
-      return PA_CHANNEL_POSITION_SIDE_LEFT;
-    case SIDE_RIGHT:
-      return PA_CHANNEL_POSITION_SIDE_RIGHT;
-    case CHANNELS_MAX:
-      return PA_CHANNEL_POSITION_INVALID;
-  }
-  NOTREACHED() << "Invalid channel " << channel;
-  return PA_CHANNEL_POSITION_INVALID;
-}
-
-static pa_channel_map ChannelLayoutToPAChannelMap(
-    ChannelLayout channel_layout) {
-  // Initialize channel map.
-  pa_channel_map channel_map;
-  pa_channel_map_init(&channel_map);
-
-  channel_map.channels = ChannelLayoutToChannelCount(channel_layout);
-
-  // All channel maps have the same size array of channel positions.
-  for (unsigned int channel = 0; channel != CHANNELS_MAX; ++channel) {
-    int channel_position = kChannelOrderings[channel_layout][channel];
-    if (channel_position > -1) {
-      channel_map.map[channel_position] = ChromiumToPAChannelPosition(
-          static_cast<Channels>(channel));
-    } else {
-      // PulseAudio expects unused channels in channel maps to be filled with
-      // PA_CHANNEL_POSITION_MONO.
-      channel_map.map[channel_position] = PA_CHANNEL_POSITION_MONO;
-    }
-  }
-
-  // Fill in the rest of the unused channels.
-  for (unsigned int channel = CHANNELS_MAX; channel != PA_CHANNELS_MAX;
-       ++channel) {
-    channel_map.map[channel] = PA_CHANNEL_POSITION_MONO;
-  }
-
-  return channel_map;
-}
-
-static size_t MicrosecondsToBytes(
-    uint32 microseconds, uint32 sample_rate, size_t bytes_per_frame) {
-  return microseconds * sample_rate * bytes_per_frame /
-      base::Time::kMicrosecondsPerSecond;
-}
-
-void PulseAudioOutputStream::ContextStateCallback(pa_context* context,
-                                                  void* state_addr) {
-  pa_context_state_t* state = static_cast<pa_context_state_t*>(state_addr);
-  *state = pa_context_get_state(context);
-}
-
-void PulseAudioOutputStream::WriteRequestCallback(
-    pa_stream* playback_handle, size_t length, void* stream_addr) {
-  PulseAudioOutputStream* stream =
-      static_cast<PulseAudioOutputStream*>(stream_addr);
-
-  DCHECK_EQ(stream->message_loop_, MessageLoop::current());
-
-  stream->write_callback_handled_ = true;
-
-  // Fulfill write request.
-  stream->FulfillWriteRequest(length);
-}
-
-PulseAudioOutputStream::PulseAudioOutputStream(const AudioParameters& params,
-                                               AudioManagerLinux* manager,
-                                               MessageLoop* message_loop)
-    : channel_layout_(params.channel_layout),
-      channel_count_(ChannelLayoutToChannelCount(channel_layout_)),
-      sample_format_(BitsToPASampleFormat(params.bits_per_sample)),
-      sample_rate_(params.sample_rate),
-      bytes_per_frame_(params.channels * params.bits_per_sample / 8),
-      manager_(manager),
-      pa_context_(NULL),
-      pa_mainloop_(NULL),
-      playback_handle_(NULL),
-      packet_size_(params.GetPacketSize()),
-      frames_per_packet_(packet_size_ / bytes_per_frame_),
-      client_buffer_(NULL),
-      volume_(1.0f),
-      stream_stopped_(true),
-      write_callback_handled_(false),
-      message_loop_(message_loop),
-      ALLOW_THIS_IN_INITIALIZER_LIST(weak_factory_(this)),
-      source_callback_(NULL) {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-  DCHECK(manager_);
-
-  // TODO(slock): Sanity check input values.
-}
-
-PulseAudioOutputStream::~PulseAudioOutputStream() {
-  // All internal structures should already have been freed in Close(),
-  // which calls AudioManagerLinux::Release which deletes this object.
-  DCHECK(!playback_handle_);
-  DCHECK(!pa_context_);
-  DCHECK(!pa_mainloop_);
-}
-
-bool PulseAudioOutputStream::Open() {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-
-  // TODO(slock): Possibly move most of this to an OpenPlaybackDevice function
-  // in a new class 'pulse_util', like alsa_util.
-
-  // Create a mainloop API and connect to the default server.
-  pa_mainloop_ = pa_mainloop_new();
-  pa_mainloop_api* pa_mainloop_api = pa_mainloop_get_api(pa_mainloop_);
-  pa_context_ = pa_context_new(pa_mainloop_api, "Chromium");
-  pa_context_state_t pa_context_state = PA_CONTEXT_UNCONNECTED;
-  pa_context_connect(pa_context_, NULL, PA_CONTEXT_NOFLAGS, NULL);
-
-  // Wait until PulseAudio is ready.
-  pa_context_set_state_callback(pa_context_, &ContextStateCallback,
-                                &pa_context_state);
-  while (pa_context_state != PA_CONTEXT_READY) {
-    pa_mainloop_iterate(pa_mainloop_, 1, NULL);
-    if (pa_context_state == PA_CONTEXT_FAILED ||
-        pa_context_state == PA_CONTEXT_TERMINATED) {
-      Reset();
-      return false;
-    }
-  }
-
-  // Set sample specifications.
-  pa_sample_spec pa_sample_specifications;
-  pa_sample_specifications.format = sample_format_;
-  pa_sample_specifications.rate = sample_rate_;
-  pa_sample_specifications.channels = channel_count_;
-
-  // Get channel mapping and open playback stream.
-  pa_channel_map* map = NULL;
-  pa_channel_map source_channel_map = ChannelLayoutToPAChannelMap(
-      channel_layout_);
-  if (source_channel_map.channels != 0) {
-    // The source data uses a supported channel map so we will use it rather
-    // than the default channel map (NULL).
-    map = &source_channel_map;
-  }
-  playback_handle_ = pa_stream_new(pa_context_, "Playback",
-                                   &pa_sample_specifications, map);
-
-  // Initialize client buffer.
-  uint32 output_packet_size = frames_per_packet_ * bytes_per_frame_;
-  client_buffer_.reset(new media::SeekableBuffer(0, output_packet_size));
-
-  // Set write callback.
-  pa_stream_set_write_callback(playback_handle_, &WriteRequestCallback, this);
-
-  // Set server-side buffer attributes.
-  // (uint32_t)-1 is the default and recommended value from PulseAudio's
-  // documentation, found at:
-  // http://freedesktop.org/software/pulseaudio/doxygen/structpa__buffer__attr.html.
-  pa_buffer_attr pa_buffer_attributes;
-  pa_buffer_attributes.maxlength = static_cast<uint32_t>(-1);
-  pa_buffer_attributes.tlength = output_packet_size;
-  pa_buffer_attributes.prebuf = static_cast<uint32_t>(-1);
-  pa_buffer_attributes.minreq = static_cast<uint32_t>(-1);
-  pa_buffer_attributes.fragsize = static_cast<uint32_t>(-1);
-
-  // Connect playback stream.
-  pa_stream_connect_playback(playback_handle_, NULL,
-                             &pa_buffer_attributes,
-                             (pa_stream_flags_t)
-                             (PA_STREAM_INTERPOLATE_TIMING |
-                             PA_STREAM_ADJUST_LATENCY |
-                             PA_STREAM_AUTO_TIMING_UPDATE),
-                             NULL, NULL);
-
-  if (!playback_handle_) {
-    Reset();
-    return false;
-  }
-
-  return true;
-}
-
-void PulseAudioOutputStream::Reset() {
-  stream_stopped_ = true;
-
-  // Close the stream.
-  if (playback_handle_) {
-    pa_stream_flush(playback_handle_, NULL, NULL);
-    pa_stream_disconnect(playback_handle_);
-
-    // Release PulseAudio structures.
-    pa_stream_unref(playback_handle_);
-    playback_handle_ = NULL;
-  }
-  if (pa_context_) {
-    pa_context_unref(pa_context_);
-    pa_context_ = NULL;
-  }
-  if (pa_mainloop_) {
-    pa_mainloop_free(pa_mainloop_);
-    pa_mainloop_ = NULL;
-  }
-
-  // Release internal buffer.
-  client_buffer_.reset();
-}
-
-void PulseAudioOutputStream::Close() {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-
-  Reset();
-
-  // Signal to the manager that we're closed and can be removed.
-  // This should be the last call in the function as it deletes "this".
-  manager_->ReleaseOutputStream(this);
-}
-
-void PulseAudioOutputStream::WaitForWriteRequest() {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-
-  if (stream_stopped_)
-    return;
-
-  // Iterate the PulseAudio mainloop.  If PulseAudio doesn't request a write,
-  // post a task to iterate the mainloop again.
-  write_callback_handled_ = false;
-  pa_mainloop_iterate(pa_mainloop_, 1, NULL);
-  if (!write_callback_handled_) {
-    message_loop_->PostTask(FROM_HERE, base::Bind(
-        &PulseAudioOutputStream::WaitForWriteRequest,
-        weak_factory_.GetWeakPtr()));
-  }
-}
-
-bool PulseAudioOutputStream::BufferPacketFromSource() {
-  uint32 buffer_delay = client_buffer_->forward_bytes();
-  pa_usec_t pa_latency_micros;
-  int negative;
-  pa_stream_get_latency(playback_handle_, &pa_latency_micros, &negative);
-  uint32 hardware_delay = MicrosecondsToBytes(pa_latency_micros,
-                                              sample_rate_,
-                                              bytes_per_frame_);
-  // TODO(slock): Deal with negative latency (negative == 1).  This has yet
-  // to happen in practice though.
-  scoped_refptr<media::DataBuffer> packet =
-      new media::DataBuffer(packet_size_);
-  size_t packet_size = RunDataCallback(packet->GetWritableData(),
-                                       packet->GetBufferSize(),
-                                       AudioBuffersState(buffer_delay,
-                                                         hardware_delay));
-
-  if (packet_size == 0)
-    return false;
-
-  media::AdjustVolume(packet->GetWritableData(),
-                      packet_size,
-                      channel_count_,
-                      bytes_per_frame_ / channel_count_,
-                      volume_);
-  packet->SetDataSize(packet_size);
-  // Add the packet to the buffer.
-  client_buffer_->Append(packet);
-  return true;
-}
-
-void PulseAudioOutputStream::FulfillWriteRequest(size_t requested_bytes) {
-  // If we have enough data to fulfill the request, we can finish the write.
-  if (stream_stopped_)
-    return;
-
-  // Request more data from the source until we can fulfill the request or
-  // fail to receive anymore data.
-  bool buffering_successful = true;
-  while (client_buffer_->forward_bytes() < requested_bytes &&
-         buffering_successful) {
-    buffering_successful = BufferPacketFromSource();
-  }
-
-  size_t bytes_written = 0;
-  if (client_buffer_->forward_bytes() > 0) {
-    // Try to fulfill the request by writing as many of the requested bytes to
-    // the stream as we can.
-    WriteToStream(requested_bytes, &bytes_written);
-  }
-
-  if (bytes_written < requested_bytes) {
-    // We weren't able to buffer enough data to fulfill the request.  Try to
-    // fulfill the rest of the request later.
-    message_loop_->PostTask(FROM_HERE, base::Bind(
-        &PulseAudioOutputStream::FulfillWriteRequest,
-        weak_factory_.GetWeakPtr(),
-        requested_bytes - bytes_written));
-  } else {
-    // Continue playback.
-    message_loop_->PostTask(FROM_HERE, base::Bind(
-        &PulseAudioOutputStream::WaitForWriteRequest,
-        weak_factory_.GetWeakPtr()));
-  }
-}
-
-void PulseAudioOutputStream::WriteToStream(size_t bytes_to_write,
-                                           size_t* bytes_written) {
-  *bytes_written = 0;
-  while (*bytes_written < bytes_to_write) {
-    const uint8* chunk;
-    size_t chunk_size;
-
-    // Stop writing if there is no more data available.
-    if (!client_buffer_->GetCurrentChunk(&chunk, &chunk_size))
-      break;
-
-    // Write data to stream.
-    pa_stream_write(playback_handle_, chunk, chunk_size,
-                    NULL, 0LL, PA_SEEK_RELATIVE);
-    client_buffer_->Seek(chunk_size);
-    *bytes_written += chunk_size;
-  }
-}
-
-void PulseAudioOutputStream::Start(AudioSourceCallback* callback) {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-
-  CHECK(callback);
-  source_callback_ = callback;
-
-  // Clear buffer, it might still have data in it.
-  client_buffer_->Clear();
-  stream_stopped_ = false;
-
-  // Start playback.
-  message_loop_->PostTask(FROM_HERE, base::Bind(
-      &PulseAudioOutputStream::WaitForWriteRequest,
-      weak_factory_.GetWeakPtr()));
-}
-
-void PulseAudioOutputStream::Stop() {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-
-  stream_stopped_ = true;
-}
-
-void PulseAudioOutputStream::SetVolume(double volume) {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-
-  volume_ = static_cast<float>(volume);
-}
-
-void PulseAudioOutputStream::GetVolume(double* volume) {
-  DCHECK_EQ(message_loop_, MessageLoop::current());
-
-  *volume = volume_;
-}
-
-uint32 PulseAudioOutputStream::RunDataCallback(
-    uint8* dest, uint32 max_size, AudioBuffersState buffers_state) {
-  if (source_callback_)
-    return source_callback_->OnMoreData(this, dest, max_size, buffers_state);
-
-  return 0;
-}