Add an ALSA output to the motown audio server.

services/media/audio/platform/linux/alsa_output.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 <set>
+
+#include "mojo/services/media/common/cpp/local_time.h"
+#include "services/media/audio/platform/linux/alsa_output.h"
+
+namespace mojo {
+namespace media {
+namespace audio {
+
+static constexpr LocalDuration TARGET_LATENCY = local_time::from_msec(35);
+static constexpr LocalDuration LOW_BUF_THRESH = local_time::from_msec(30);
+static constexpr LocalDuration ERROR_RECOVERY_TIME = local_time::from_msec(300);
+static constexpr LocalDuration WAIT_FOR_ALSA_DELAY = local_time::from_usec(500);
+static const std::set<uint8_t> SUPPORTED_CHANNEL_COUNTS({ 1, 2 });
+static const std::set<uint32_t> SUPPORTED_SAMPLE_RATES({
+    48000, 32000, 24000, 16000, 8000, 4000,
+    44100, 22050, 11025,
+});
+
+static inline bool IsRecoverableAlsaError(int error_code) {
+  switch (error_code) {
+  case -EINTR:
+  case -EPIPE:
+  case -ESTRPIPE:
+    return true;
+  default:
+    return false;
+  }
+}
+
+AudioOutputPtr CreateDefaultAlsaOutput(AudioOutputManager* manager) {
+  // TODO(johngro): Do better than this.  If we really want to support
+  // Linux/ALSA as a platform, we should be creating one output for each
+  // physical output in the system, matching our configuration to the physical
+  // output's configuration, and disabling resampling at the ALSA level.
+  //
+  // If we could own the output entirely and bypass the mixer to achieve lower
+  // latency, that would be even better.
+  AudioOutputPtr audio_out(audio::AlsaOutput::New(manager));
+  if (!audio_out) { return nullptr; }
+
+  AlsaOutput* alsa_out = static_cast<AlsaOutput*>(audio_out.get());
+  DCHECK(alsa_out);
+
+  LpcmMediaTypeDetailsPtr config(LpcmMediaTypeDetails::New());
+  config->frames_per_second = 48000;
+  config->samples_per_frame = 2;
+  config->sample_format = LpcmSampleFormat::SIGNED_16;
+
+  if (alsa_out->Configure(config.Pass()) != MediaResult::OK) {
+    return nullptr;
+  }
+
+  return audio_out;
+}
+
+AlsaOutput::AlsaOutput(AudioOutputManager* manager)
+  : StandardOutputBase(manager) {}
+
+AlsaOutput::~AlsaOutput() {
+  // We should have been cleaned up already, but in release builds, call cleanup
+  // anyway, just in case something got missed.
+  DCHECK(!alsa_device_);
+  Cleanup();
+}
+
+AudioOutputPtr AlsaOutput::New(AudioOutputManager* manager) {
+  return AudioOutputPtr(new AlsaOutput(manager));
+}
+
+MediaResult AlsaOutput::Configure(LpcmMediaTypeDetailsPtr config) {
+  if (!config) { return MediaResult::INVALID_ARGUMENT; }
+  if (output_format_) { return MediaResult::BAD_STATE; }
+
+  uint32_t bytes_per_sample;
+  switch (config->sample_format) {
+  case LpcmSampleFormat::UNSIGNED_8:
+    alsa_format_ = SND_PCM_FORMAT_U8;
+    silence_byte_ = 0x80;
+    bytes_per_sample = 1;
+    break;
+
+  case LpcmSampleFormat::SIGNED_16:
+    alsa_format_ = SND_PCM_FORMAT_S16;
+    silence_byte_ = 0x00;
+    bytes_per_sample = 2;
+    break;
+
+  case LpcmSampleFormat::SIGNED_24_IN_32:
+  default:
+    return MediaResult::UNSUPPORTED_CONFIG;
+  }
+
+  if (SUPPORTED_SAMPLE_RATES.find(config->frames_per_second) ==
+      SUPPORTED_SAMPLE_RATES.end()) {
+    return MediaResult::UNSUPPORTED_CONFIG;
+  }
+
+  if (SUPPORTED_CHANNEL_COUNTS.find(config->samples_per_frame) ==
+      SUPPORTED_CHANNEL_COUNTS.end()) {
+    return MediaResult::UNSUPPORTED_CONFIG;
+  }
+
+  // Compute the ratio between frames and local time ticks.
+  LinearTransform::Ratio sec_per_tick(LocalDuration::period::num,
+                                      LocalDuration::period::den);
+  LinearTransform::Ratio frames_per_sec(config->frames_per_second, 1);
+  bool is_precise = LinearTransform::Ratio::Compose(frames_per_sec,
+                                                    sec_per_tick,
+                                                    &frames_per_tick_);
+  DCHECK(is_precise);
+
+  // Figure out how many bytes there are per frame.
+  output_bytes_per_frame_ = bytes_per_sample * config->samples_per_frame;
+
+  // Success
+  output_format_ = config.Pass();
+  return MediaResult::OK;
+}
+
+MediaResult AlsaOutput::Init() {
+  static const char* kAlsaDevice = "default";
+
+  if (!output_format_) { return MediaResult::BAD_STATE; }
+  if (alsa_device_) { return MediaResult::BAD_STATE; }
+
+  snd_pcm_sframes_t res;
+  res = snd_pcm_open(&alsa_device_,
+                     kAlsaDevice,
+                     SND_PCM_STREAM_PLAYBACK,
+                     SND_PCM_NONBLOCK);
+  if (res != 0) {
+    LOG(ERROR) << "Failed to open ALSA device \"" << kAlsaDevice << "\".";
+    return MediaResult::INTERNAL_ERROR;
+  }
+
+  res = snd_pcm_set_params(alsa_device_,
+                           alsa_format_,
+                           SND_PCM_ACCESS_RW_INTERLEAVED,
+                           output_format_->samples_per_frame,
+                           output_format_->frames_per_second,
+                           0,   // do not allow ALSA resample
+                           local_time::to_usec<unsigned int>(TARGET_LATENCY));
+  if (res) {
+    LOG(ERROR) << "Failed to configure ALSA device \"" << kAlsaDevice << "\" "
+               << "(res = " << res << ")";
+    LOG(ERROR) << "Requested samples per frame: "
+               << output_format_->samples_per_frame;
+    LOG(ERROR) << "Requested frames per second: "
+               << output_format_->frames_per_second;
+    LOG(ERROR) << "Requested ALSA format      : " << alsa_format_;
+    Cleanup();
+    return MediaResult::INTERNAL_ERROR;
+  }
+
+  // Figure out how big our mixing buffer needs to be, then allocate it.
+  res = snd_pcm_avail_update(alsa_device_);
+  if (res <= 0) {
+    LOG(ERROR) << "[" << this << "] : "
+               << "Fatal error (" << res
+               << ") attempting to determine ALSA buffer size.";
+    Cleanup();
+    return MediaResult::INTERNAL_ERROR;
+  }
+
+  mix_buf_frames_ = res;
+  mix_buf_.reset(new uint8_t[mix_buf_frames_ * output_bytes_per_frame_]);
+
+  return MediaResult::OK;
+}
+
+void AlsaOutput::Cleanup() {
+  if (alsa_device_) {
+    snd_pcm_close(alsa_device_);
+    alsa_device_ = nullptr;
+  }
+
+  mix_buf_ = nullptr;
+  mix_buf_frames_ = 0;
+}
+
+bool AlsaOutput::StartMixJob(MixJob* job, const LocalTime& process_start) {
+  DCHECK(job);
+
+  // Are we not primed?  If so, fill a mix buffer with silence and send it to
+  // the alsa device.  Schedule a callback for a short time in the future so
+  // ALSA has a chance to start the output and we can take our best guess of the
+  // function which maps output frames to local time.
+  if (!primed_) {
+    HandleAsUnderflow();
+    return false;
+  }
+
+  // Figure out how many frames of audio we need to produce in order to top off
+  // the buffer.  If we are primed, but do not know the transformation between
+  // audio frames and local time ticks, do our best to figure it out in the
+  // process.
+  snd_pcm_sframes_t avail;
+  if (!local_to_output_known_) {
+    snd_pcm_sframes_t delay;
+
+    int res = snd_pcm_avail_delay(alsa_device_, &avail, &delay);
+    LocalTime now = LocalClock::now();
+
+    if (res < 0) {
+      HandleAlsaError(res);
+      return false;
+    }
+
+    DCHECK_GE(delay, 0);
+    int64_t now_ticks = now.time_since_epoch().count();
+    local_to_output_ = LinearTransform(now_ticks, frames_per_tick_, -delay);
+    local_to_output_known_ = true;
+    frames_sent_ = 0;
+    while (++local_to_output_gen_ == MixJob::INVALID_GENERATION) {}
+  } else {
+    avail = snd_pcm_avail_update(alsa_device_);
+    if (avail < 0) {
+      HandleAlsaError(avail);
+      return false;
+    }
+  }
+
+  // Compute the time that we think we will completely underflow, then back off
+  // from that by the low buffer threshold and use that to determine when we
+  // should mix again.
+  int64_t playout_time_ticks;
+  bool trans_ok = local_to_output_.DoReverseTransform(frames_sent_,
+                                                      &playout_time_ticks);
+  DCHECK(trans_ok);
+  LocalTime playout_time = LocalTime(LocalDuration(playout_time_ticks));
+  LocalTime low_buf_time = playout_time - LOW_BUF_THRESH;
+
+  if (process_start >= low_buf_time) {
+    // Because of the way that ALSA consumes data and updates its internal
+    // bookkeeping, it is possible that we are past our low buffer threshold,
+    // but ALSA still thinks that there is no room to write new frames.  If this
+    // is the case, just try again a short amount of time in the future.
+    DCHECK_GE(avail, 0);
+    if (!avail) {
+      SetNextSchedDelay(WAIT_FOR_ALSA_DELAY);
+      return false;
+    }
+
+    // Limit the amt that we queue to be no more than what ALSA will currently
+    // accept, or what it currently will take to fill us to our target latency.
+    //
+    // The playout target had better be ahead of the playout time, or we are
+    // almost certainly going to underflow.  If this happens, for whatever
+    // reason, just try to send a full buffer and deal with the underflow when
+    // ALSA notices it.
+    int64_t fill_amt;
+    LocalTime playout_target = LocalClock::now() + TARGET_LATENCY;
+    if (playout_target > playout_time) {
+      fill_amt = (playout_target - playout_time).count();
+    } else {
+      fill_amt = TARGET_LATENCY.count();
+    }
+
+    DCHECK_GE(fill_amt, 0);
+    DCHECK_LE(fill_amt, std::numeric_limits<int32_t>::max());
+    fill_amt *= frames_per_tick_.numerator;
+    fill_amt += frames_per_tick_.denominator - 1;
+    fill_amt /= frames_per_tick_.denominator;
+
+    job->buf_frames = (avail < fill_amt) ? avail : fill_amt;
+    if (job->buf_frames > mix_buf_frames_) {
+      job->buf_frames = mix_buf_frames_;
+    }
+
+    job->buf = mix_buf_.get();
+    job->start_pts_of = frames_sent_;
+    job->local_to_output = &local_to_output_;
+    job->local_to_output_gen = local_to_output_gen_;
+
+    // TODO(johngro): optimize this if we can.  The first buffer we mix can just
+    // put its samples directly into the output buffer, and does not need to
+    // accumulate and clip.  In theory, we only need to put silence in the
+    // places where our outputs are not going to already overwrite.
+    FillMixBufWithSilence(job->buf_frames);
+    return true;
+  }
+
+  // Wait until its time to mix some more data.
+  SetNextSchedTime(low_buf_time);
+  return false;
+}
+
+bool AlsaOutput::FinishMixJob(const MixJob& job) {
+  DCHECK(job.buf == mix_buf_.get());
+  DCHECK(job.buf_frames);
+
+  // We should always be able to write all of the data that we mixed.
+  snd_pcm_sframes_t res;
+  res = snd_pcm_writei(alsa_device_, job.buf, job.buf_frames);
+  if (res != job.buf_frames) {
+    HandleAlsaError(res);
+    return false;
+  }
+
+  frames_sent_ += res;
+  return true;
+}
+
+void AlsaOutput::FillMixBufWithSilence(uint32_t frames) {
+  DCHECK(mix_buf_);
+  DCHECK(frames <= mix_buf_frames_);
+
+  // TODO(johngro): someday, this may not be this simple.  Filling unsigned
+  // multibyte sample formats, or floating point formats, will require something
+  // more sophisticated than filling with a single byte pattern.
+  ::memset(mix_buf_.get(), silence_byte_, frames * output_bytes_per_frame_);
+}
+
+void AlsaOutput::HandleAsUnderflow() {
+  snd_pcm_sframes_t res;
+
+  // If we were already primed, then this is a legitimate underflow, not the
+  // startup case or recovery from some other error.
+  if (primed_) {
+    // TODO(johngro): come up with a way to properly throttle this.  Also, add a
+    // friendly name to the output so the log helps to identify which output
+    // underflowed.
+    LOG(WARNING) << "[" << this << "] : underflow";
+    res = snd_pcm_recover(alsa_device_, -EPIPE, true);
+    if (res < 0) {
+      HandleAsError(res);
+      return;
+    }
+  }
+
+  // TODO(johngro): We don't actually have to fill up the entire lead time with
+  // silence.  When we have better control of our thread priorities, prime this
+  // with the minimimum amt we can get away with and still be able to start
+  // mixing without underflowing.
+  FillMixBufWithSilence(mix_buf_frames_);
+  res = snd_pcm_writei(alsa_device_, mix_buf_.get(), mix_buf_frames_);
+
+  if (res < 0) {
+    HandleAsError(res);
+    return;
+  }
+
+  primed_ = true;
+  local_to_output_known_ = false;
+  SetNextSchedDelay(local_time::from_msec(1));
+}
+
+void AlsaOutput::HandleAsError(snd_pcm_sframes_t code) {
+  // TODO(johngro): Throttle this somehow.
+  LOG(WARNING) << "[" << this << "] : Attempting to recover from ALSA error "
+               << code;
+
+  if (IsRecoverableAlsaError(code)) {
+    snd_pcm_sframes_t new_code;
+
+    new_code = snd_pcm_recover(alsa_device_, code, true);
+    DCHECK(!new_code || (new_code == code));
+
+    // If we recovered, or we didn't and the original error was EINTR, schedule
+    // a retry time in the future and unwind.
+    //
+    // TODO(johngro): revisit the topic of errors we fail to snd_pcm_recover
+    // from.  If we cannot recover from them, we should probably close and
+    // re-open the device.  No matter what, we should put some form of limit on
+    // how many times we try before really giving up and shutting down the
+    // output for good.  We also need to invent a good way to test these edge
+    // cases.
+    if (!new_code || (new_code == -EINTR)) {
+      primed_ = false;
+      local_to_output_known_ = false;
+      SetNextSchedDelay(ERROR_RECOVERY_TIME);
+    }
+  }
+
+  LOG(ERROR) << "[" << this << "] : Fatal ALSA error "
+             << code << ".  Shutting down";
+  ShutdownSelf();
+}
+
+void AlsaOutput::HandleAlsaError(snd_pcm_sframes_t code) {
+  // ALSA signals an underflow by returning -EPIPE from jobs.  If the error code
+  // is -EPIPE, treat this as an underflow and attempt to reprime the pipeline.
+  if (code == -EPIPE) {
+    HandleAsUnderflow();
+  } else {
+    HandleAsError(code);
+  }
+}
+
+}  // namespace audio
+}  // namespace media
+}  // namespace mojo
+