PulseAudio Sound Playback on Linux

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
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index 0000000000000000000000000000000000000000..af501d05b9bcaf67a601043be1f37427245fe82b
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+++ b/media/audio/linux/pulse_output.cc
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+// 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 "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 LSBs of 32-bit field, little endian),
+    //          and PA_SAMPLE_24_32BE (in LSBs 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* userdata) {
+  int* context_ready = static_cast<int*>(userdata);

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

Change int* to pa_context_state_t*, then just do

*context_ready = pa_context_get_state(context) (no need for switch statement
below)

[slock, 2011/08/10 22:41:04]

Done.

+  pa_context_state_t state = pa_context_get_state(context);
+  switch(state) {
+    default:
+      break;
+    case PA_CONTEXT_FAILED:
+        *context_ready = 3;
+    case PA_CONTEXT_TERMINATED:
+        *context_ready = 2;
+    case PA_CONTEXT_READY:
+        *context_ready = 0;
+  }
+}
+
+void PulseAudioOutputStream::WriteCallback(pa_stream* stream, size_t length,
+                                           void* userdata) {
+  PulseAudioOutputStream* stream_ptr =
+      static_cast<PulseAudioOutputStream*>(userdata);
+
+  // Request data from upstream if necessary.
+  while (stream_ptr->client_buffer_->forward_bytes() < length &&
+         !stream_ptr->source_exhausted_) {
+    stream_ptr->BufferPacketInClient();
+  }
+
+  // Get data to write.
+  scoped_array<uint8> read_data(new uint8[length]);
+  stream_ptr->client_buffer_->Read(read_data.get(), length);
+  // Write to stream.
+  pa_stream_write(stream, read_data.get(), length, NULL, 0LL, PA_SEEK_RELATIVE);
+}
+
+PulseAudioOutputStream::PulseAudioOutputStream(const AudioParameters& params,
+                                               AudioManagerLinux* manager)
+      : channel_layout_(params.channel_layout),
+        sample_format_(BitsToFormat(params.bits_per_sample)),
+        sample_rate_(params.sample_rate),
+        bytes_per_frame_(params.channels * params.bits_per_sample / 8),
+        packet_size_(params.GetPacketSize()),
+        frames_per_packet_(packet_size_ / bytes_per_frame_),
+        stream_stopped_(false),
+        manager_(manager),
+        pa_mainloop_(NULL),
+        pa_mainloop_api_(NULL),
+        pa_context_(NULL),
+        playback_handle_(NULL),
+        client_buffer_(NULL),
+        source_exhausted_(false),
+        source_callback_(NULL) {
+  // TODO(slock): Sanity check input values.
+}
+
+PulseAudioOutputStream::~PulseAudioOutputStream() {
+  // All internal structures are already freed in Close(), which calls
+  // AudioManagerLinux::Release which deletes this object.

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

DCHECK playback_handle_, pa_context_, pa_mainloop_ are all NULL

[slock, 2011/08/10 22:41:04]

Done.

+}
+
+bool PulseAudioOutputStream::Open() {
+  // TODO(slock): Possibly move most of this to a 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_get_api(pa_mainloop_);
+  pa_context_ = pa_context_new(pa_mainloop_api_, "Chromium");
+  pa_context_connect(pa_context_, NULL, PA_CONTEXT_NOFLAGS, NULL);
+
+  // Wait until PulseAudio is ready.
+  int pa_context_ready = 1;

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

(See comment in ContextStateCallback)

Change int to pa_context_state_t.

[slock, 2011/08/10 22:41:04]

Done.

+  pa_context_set_state_callback(pa_context_, &ContextStateCallback,
+                                &pa_context_ready);
+  while (pa_context_ready == 1)
+    pa_mainloop_iterate(pa_mainloop_, 1, NULL);
+  if (pa_context_ready != 0) {

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

if (pa_context_ready != PA_CONTEXT_READY)

[slock, 2011/08/10 22:41:04]

Done.

+    stream_stopped_ = false;

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

remove line

[slock, 2011/08/10 22:41:04]

Done.

+    return false;
+  }
+
+  // Set sample specifications and open playback stream.
+  pa_sample_specs_.format = sample_format_;
+  pa_sample_specs_.rate = sample_rate_;
+  pa_sample_specs_.channels = ChannelLayoutToChannelCount(channel_layout_);
+  playback_handle_ = pa_stream_new(pa_context_, "Playback",
+                                   &pa_sample_specs_, 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_, &WriteCallback, this);
+
+  // Set server side buffer attributes.
+  // TODO(slock): Figure out what these values should actually be, for now use
+  // recommended values from PulseAudio's documentation:
+  // http://freedesktop.org/software/pulseaudio/doxygen/structpa__buffer__attr.html
+  pa_buffer_attributes_.maxlength = (uint32_t)-1;
+  pa_buffer_attributes_.tlength = output_packet_size;
+  pa_buffer_attributes_.prebuf = (uint32_t)-1;
+  pa_buffer_attributes_.minreq = (uint32_t)-1;
+  pa_buffer_attributes_.fragsize = (uint32_t)-1;
+
+  // Set volume

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

nit: complete sentence

[slock, 2011/08/10 22:41:04]

Done.

+  pa_volume_.channels = ChannelLayoutToChannelCount(channel_layout_);
+  pa_cvolume_set(&pa_volume_, pa_volume_.channels, PA_VOLUME_NORM);
+
+  // 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),
+                             &pa_volume_, NULL);
+
+  if (!playback_handle_) {
+    stream_stopped_ = true;
+    return false;
+  }
+
+  return true;
+}
+
+void PulseAudioOutputStream::Close() {
+/*

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

commented out?

[slock, 2011/08/10 22:41:04]

Done.

+  // Close the device.
+  if (playback_handle_) {

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

Set playback_handle_, pa_context_, pa_mainloop_ to NULL after freeing each of
them.

[slock, 2011/08/10 22:41:04]

Done.

+    pa_stream_flush(playback_handle_, NULL, NULL);
+    pa_stream_disconnect(playback_handle_);
+
+    // Release PulseAudio structures.
+    pa_stream_unref(playback_handle_);
+  }
+  if (pa_context_)
+    pa_context_unref(pa_context_);
+  if (pa_mainloop_)
+    pa_mainloop_free(pa_mainloop_);
+
+  // Release internal buffer.
+  client_buffer_.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::BufferPacketInClient() {
+  // Request more data if we have more capacity.
+  if (client_buffer_->forward_capacity() > client_buffer_->forward_bytes()) {
+
+    // Before making request to source for data we need to determine the delay
+    // (in bytes) for the requested data to be played.
+    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));
+    CHECK(packet_size <= packet->GetBufferSize()) <<
+        "Data source overran buffer.";
+
+    // TODO(slock): Swizzling, downmixing, and volume adjusting.
+
+    if (packet_size > 0) {
+      packet->SetDataSize(packet_size);
+      // Add the packet to the buffer.
+      client_buffer_->Append(packet);
+    } else {
+      source_exhausted_ = true;
+    }
+  }
+}
+
+void PulseAudioOutputStream::ClientBufferLoop() {
+  while(!stream_stopped_ && !source_exhausted_) {
+    // As long as the stream is active, we should be buffering packets if need

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

Write comment outside of the while loop and reword. I think all you really need
to mention is pa_mainloop_iterate blocks until PulseAudio is ready to write
audio data, at which point WirteCallback is called (on the same thread in which
PulseAudioOutputStream is created)

[slock, 2011/08/10 22:41:04]

Done.

+    // be and writing packets if need be.  These are asynchronous processes.
+    // This loop buffers packets and the PulseAudio mainloop writes them.
+    // BufferPacket() only actually buffers under certain circumstances and
+    // pa_mainloop_iterate() only calls WriteCallback under certain
+    // circumstances, but the loop marches on in either case.
+    pa_mainloop_iterate(pa_mainloop_, 1, NULL);
+  }
+}
+
+void PulseAudioOutputStream::Start(AudioSourceCallback* callback) {
+  CHECK(callback);
+  source_callback_ = callback;
+
+  // Clear buffer, it might still have data in it.
+  client_buffer_->Clear();
+  source_exhausted_ = false;
+
+  // Start playing.
+  ClientBufferLoop();

[vrk (LEFT CHROMIUM), 2011/08/10 16:34:51]

inline function

[slock, 2011/08/10 22:41:04]

Done.

+}
+
+void PulseAudioOutputStream::Stop() {
+  // Effect will not be instantaneous as the PulseAudio server buffer drains.
+  // TODO(slock): Immediate stopping.
+  stream_stopped_ = true;
+}
+
+void PulseAudioOutputStream::SetVolume(double volume) {
+  pa_volume_t new_volume = pa_sw_volume_from_linear(volume);
+  pa_cvolume_set(&pa_volume_, pa_volume_.channels, new_volume);
+}
+
+void PulseAudioOutputStream::GetVolume(double* volume) {
+  // We do not allow volume changes on a per-channel basis, so all channels will
+  // always have the same volume and the average will reflect this.
+  *volume = pa_cvolume_avg(&pa_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;
+}