PulseAudio Sound Playback on Linux

media/audio/linux/pulse_output.cc

+// 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/message_loop.h"
+#include "media/audio/audio_util.h"
+#include "media/audio/audio_parameters.h"

[vrk (LEFT CHROMIUM), 2011/08/18 18:22:22]

nit: abc order of includes

[slock, 2011/08/18 18:54:16]

Done.

+#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 BitsToFormat(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 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);
+
+  // Continue playback.
+  stream->message_loop_->PostTask(

[vrk (LEFT CHROMIUM), 2011/08/18 18:22:22]

I believe this should be in the "else" block for if (bytes_written <
requested_bytes) in FulfillWriteRequest.

(If you haven't fulfilled the outstanding request, post task to get more data.
Else, post task to process more PA callbacks.)

[slock, 2011/08/18 18:54:16]

Done.

+      FROM_HERE,
+      stream->method_factory_.NewRunnableMethod(
+          &PulseAudioOutputStream::WaitForWriteRequest));
+}
+
+PulseAudioOutputStream::PulseAudioOutputStream(const AudioParameters& params,
+                                               AudioManagerLinux* manager,
+                                               MessageLoop* message_loop)
+    : channel_layout_(params.channel_layout),
+      channel_count_(ChannelLayoutToChannelCount(channel_layout_)),
+      sample_format_(BitsToFormat(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(method_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 a OpenPlaybackDevice function in

[vrk (LEFT CHROMIUM), 2011/08/18 18:22:22]

nit: "to an"

[slock, 2011/08/18 18:54:16]

Done.

+  // 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 and open playback stream.
+  pa_sample_spec pa_sample_specifications;
+  pa_sample_specifications.format = sample_format_;
+  pa_sample_specifications.rate = sample_rate_;
+  pa_sample_specifications.channels = channel_count_;
+  playback_handle_ = pa_stream_new(pa_context_, "Playback",
+                                   &pa_sample_specifications, NULL);
+
+  // 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);

[vrk (LEFT CHROMIUM), 2011/08/18 18:22:22]

nit: move this to line above?

[slock, 2011/08/18 18:54:16]

Done.

+
+  // 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());
+
+  // Iterate the PulseAudio mainloop.  If the stream isn't stopped or 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_ && !stream_stopped_) {
+    message_loop_->PostTask(
+        FROM_HERE,
+        method_factory_.NewRunnableMethod(
+            &PulseAudioOutputStream::WaitForWriteRequest));
+  }
+}
+
+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;
+
+  // TODO(slock): Swizzling and downmixing.
+  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

[vrk (LEFT CHROMIUM), 2011/08/18 18:22:22]

nit: Put the comment above the if statement, or put curly braces around the if
statement (must have {} around multi-line blocks)

[slock, 2011/08/18 18:54:16]

Done.  I didn't if 'multi-line' included comments and couldn't find it in the
style-guide.

+    // the stream as we can.
+    WriteToStream(requested_bytes, &bytes_written);
+
+  if (bytes_written < requested_bytes)

[vrk (LEFT CHROMIUM), 2011/08/18 18:22:22]

nit: curly braces around multi-line blocks

[slock, 2011/08/18 18:54:16]

Done.

+    // 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,
+        method_factory_.NewRunnableMethod(
+            &PulseAudioOutputStream::FulfillWriteRequest,
+            requested_bytes - bytes_written));
+}
+
+void PulseAudioOutputStream::WriteToStream(size_t bytes_to_write,
+                                           size_t* bytes_written) {
+  *bytes_written = 0;
+  const uint8* chunk;
+  size_t chunk_size;
+
+  // Try to get data to write.
+  bool buffer_exhausted = !client_buffer_->GetCurrentChunk(&chunk, &chunk_size);
+  while (*bytes_written < bytes_to_write && !buffer_exhausted) {
+    // 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;
+
+    // Get more data.
+    buffer_exhausted = client_buffer_->GetCurrentChunk(&chunk, &chunk_size);

[vrk (LEFT CHROMIUM), 2011/08/18 18:22:22]

logic is incorrect: should be 
buffer_exhausted = !client_buffer_->GetCurrentChunk(&chunk, &chunk_size);


In fact, instead of using the buffer_exhausted variable, perhaps a cleaner way
to write this loop would be:

while (*bytes_written < bytes_to_write) {
  const uint8* chunk;
  size_t chunk_size;
  // Stop writing if there is no more data buffered.
  if (!client_buffer_->GetCurrentChunk(&chunk, &chunk_size))
    break;
  ...
}

[slock, 2011/08/18 18:54:16]

Done.

+  }
+}
+
+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,
+      method_factory_.NewRunnableMethod(
+          &PulseAudioOutputStream::WaitForWriteRequest));
+}
+
+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;
+}