media/audio/linux: Add CrasInputStream.

media/audio/linux/cras_input.cc

+// Copyright (c) 2012 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/cras_input.h"
+
+#include "base/basictypes.h"
+#include "base/bind.h"
+#include "base/logging.h"
+#include "base/time.h"
+#include "media/audio/audio_manager.h"
+#include "media/audio/linux/alsa_util.h"
+#include "media/audio/linux/audio_manager_linux.h"
+
+namespace media {
+
+CrasInputStream::CrasInputStream(const AudioParameters& params,
+                                 AudioManagerLinux* manager)
+    : audio_manager_(manager),
+      bytes_per_frame_(0),
+      callback_(NULL),
+      client_(NULL),
+      frame_rate_(params.sample_rate()),

[no longer working on chromium, 2012/06/20 10:45:26]

How about changing the name to sample_rate_? 
We definitely want using the same name for all the implementations.

[dgreid, 2012/06/20 17:14:25]

Done.

+      num_channels_(params.channels()),
+      params_(params),
+      pcm_format_(alsa_util::BitsToFormat(params.bits_per_sample())),
+      stream_id_(0) {
+  DCHECK(audio_manager_);
+}
+
+CrasInputStream::~CrasInputStream() {}

[no longer working on chromium, 2012/06/20 10:45:26]

Add a DCHECK here to make sure things have already cleaned up.

[dgreid, 2012/06/20 17:14:25]

Done.

+
+bool CrasInputStream::Open() {
+  if (client_) {
+    NOTREACHED() << "CrasInputStream already open";
+    return false;  // Already open.
+  }
+
+  // Sanity check input values.
+  if (params_.sample_rate() <= 0) {
+    LOG(WARNING) << "Unsupported audio frequency.";

[no longer working on chromium, 2012/06/20 10:45:26]

I know we use LOG for alsa too, but is it really needed? What about using DLOG
instead.

[dgreid, 2012/06/20 17:14:25]

Done for whole file.

+    return false;
+  }
+
+  if (AudioParameters::AUDIO_PCM_LINEAR != params_.format() &&
+      AudioParameters::AUDIO_PCM_LOW_LATENCY != params_.format()) {
+    LOG(WARNING) << "Unsupported audio format.";

[no longer working on chromium, 2012/06/20 10:45:26]

ditto

+    return false;
+  }
+
+  if (pcm_format_ == SND_PCM_FORMAT_UNKNOWN) {
+    LOG(WARNING) << "Unsupported bits/sample: " << params_.bits_per_sample();

[no longer working on chromium, 2012/06/20 10:45:26]

ditto

+    return false;
+  }
+
+  // Create the client and connect to the CRAS server.
+  if (cras_client_create(&client_) < 0) {
+    LOG(WARNING) << "Couldn't create CRAS client.\n";

[no longer working on chromium, 2012/06/20 10:45:26]

ditto

+    client_ = NULL;
+    return false;
+  }
+
+  if (cras_client_connect(client_)) {
+    LOG(WARNING) << "Couldn't connect CRAS client.\n";

[no longer working on chromium, 2012/06/20 10:45:26]

ditto

+    cras_client_destroy(client_);
+    client_ = NULL;
+    return false;
+  }
+
+  // Then start running the client.
+  if (cras_client_run_thread(client_)) {
+    LOG(WARNING) << "Couldn't run CRAS client.\n";
+    cras_client_destroy(client_);
+    client_ = NULL;
+    return false;
+  }
+
+  return true;
+}
+
+void CrasInputStream::Close() {
+  if (client_) {
+    cras_client_stop(client_);
+    cras_client_destroy(client_);
+    client_ = NULL;
+  }
+
+  if (callback_)
+    callback_->OnClose(this);
+
+  // Signal to the manager that we're closed and can be removed.
+  // Should be last call in the method as it deletes "this".
+  audio_manager_->ReleaseInputStream(this);
+}
+
+void CrasInputStream::Start(AudioInputCallback* callback) {
+  DCHECK(client_);
+  DCHECK(!callback_ && callback);

[no longer working on chromium, 2012/06/20 10:45:26]

Does this mean we only allow calling Start() once?
What about if the users want to pause/start the stream?

[dgreid, 2012/06/20 17:14:25]

Yes, call Start once until Stop is called.  This struck me as odd as well. It is
the same restriction in alsa_input.  Is there a need to call it twice?

+  callback_ = callback;
+
+  // Prepare |audio_format| and |stream_params| for the stream we
+  // will create.
+  cras_audio_format* audio_format = cras_audio_format_create(
+      pcm_format_,
+      frame_rate_,
+      num_channels_);
+  if (!audio_format) {
+    LOG(WARNING) << "Error setting up audio parameters.";
+    callback->OnError(this, -ENOMEM);
+    return;
+  }
+
+  unsigned int frames_per_packet = params_.frames_per_buffer();
+  cras_stream_params* stream_params = cras_client_stream_params_create(
+      CRAS_STREAM_INPUT,
+      frames_per_packet,  // Total latency.

[no longer working on chromium, 2012/06/20 10:45:26]

Curiously, does it mean we are using one queue with the size of
frames_per_packet as buffer?

[dgreid, 2012/06/20 17:14:25]

On the server side there are two buffers of this size created.  They are passed
to the capture callback ping-pong style.

+      frames_per_packet,  // Call back when this many ready.
+      frames_per_packet,  // Don't care for capture.

[no longer working on chromium, 2012/06/20 10:45:26]

I am curious, what does this parameter mean?

[dgreid, 2012/06/20 17:14:25]

For an output stream, it indicates the minimum amount of frames the client wants
to send at a time.  For capture streams it is ignored, comment updated.

+      CRAS_STREAM_TYPE_DEFAULT,
+      0,  // Unused flags.
+      this,
+      CrasInputStream::SamplesReady,
+      CrasInputStream::StreamError,
+      audio_format);
+  if (!stream_params) {
+    LOG(WARNING) << "Error setting up stream parameters.";
+    callback->OnError(this, -ENOMEM);
+    cras_audio_format_destroy(audio_format);
+    return;
+  }
+
+  // Before starting the stream, save the number of bytes in a frame for use in
+  // the callback.
+  bytes_per_frame_ = cras_client_format_bytes_per_frame(audio_format);
+
+  // Adding the stream will start the audio callbacks.
+  if (cras_client_add_stream(client_, &stream_id_, stream_params) == 0) {
+    audio_manager_->IncreaseActiveInputStreamCount();

[no longer working on chromium, 2012/06/20 10:45:26]

In case Start() fails for any of the reason, should users call stop()?
We call DecreaseActiveInputStreamCount() in Stop() as long as !client ||
!callback_, this does not match what Start() does.

[dgreid, 2012/06/20 17:14:25]

Thanks, good catch.  Made them match up, so that client_ will only be non-null
if IncreaseActiveInputStreamCount() was called.

+  } else {
+    LOG(WARNING) << "Failed to add the stream";

[no longer working on chromium, 2012/06/20 10:45:26]

DLOG, and add a period.

[dgreid, 2012/06/20 17:14:25]

Done.

+    callback->OnError(this, -EIO);
+  }
+
+  // Done with config params.
+  cras_audio_format_destroy(audio_format);
+  cras_client_stream_params_destroy(stream_params);
+}
+
+void CrasInputStream::Stop() {
+  if (!client_ || !callback_)
+    return;
+
+  // Removing the stream from the client stops audio.
+  cras_client_rm_stream(client_, stream_id_);
+
+  audio_manager_->DecreaseActiveInputStreamCount();
+}
+
+// Static callback asking for samples.  Run on high priority thread.
+int CrasInputStream::SamplesReady(cras_client* client,
+                                  cras_stream_id_t stream_id,
+                                  uint8* samples,
+                                  size_t frames,
+                                  const timespec* sample_ts,
+                                  void* arg) {
+  CrasInputStream* me = static_cast<CrasInputStream*>(arg);
+  me->ReadAudio(frames, samples, sample_ts);
+  return frames;
+}
+
+// Static callback for stream errors.
+int CrasInputStream::StreamError(cras_client* client,
+                                 cras_stream_id_t stream_id,
+                                 int err,
+                                 void* arg) {
+  CrasInputStream* me = static_cast<CrasInputStream*>(arg);
+  me->NotifyStreamError(err);
+  return 0;
+}
+
+void CrasInputStream::ReadAudio(size_t frames,
+                                uint8* buffer,
+                                const timespec* sample_ts) {
+  DCHECK(callback_);
+
+  timespec latency_ts  = {0, 0};
+
+  // Determine latency and pass that on to the sink.  sample_ts is the wall time
+  // indicating when the first sample in the buffer was captured.  Convert that
+  // to latency in bytes.
+  cras_client_calc_capture_latency(sample_ts, &latency_ts);
+  double latency_usec = (latency_ts.tv_sec * 1e6) + latency_ts.tv_nsec * 1e-3;
+  double frames_latency = latency_usec * frame_rate_ * 1e-6;
+  unsigned int bytes_latency =
+      static_cast<unsigned int>(frames_latency * bytes_per_frame_);
+
+  // Update the AGC volume level once every second. Note that, |volume| is
+  // also updated each time SetVolume() is called through IPC by the
+  // render-side AGC.
+  double normalized_volume = 0.0;
+  QueryAgcVolume(&normalized_volume);
+
+  callback_->OnData(this,
+                    buffer,
+                    frames * bytes_per_frame_,
+                    bytes_latency,
+                    normalized_volume);
+}
+
+void CrasInputStream::NotifyStreamError(int err) {
+  if (callback_)
+    callback_->OnError(this, err);
+}
+
+double CrasInputStream::GetMaxVolume() {
+  DCHECK(client_);
+
+  // Return the max volume as a double to match what the alsa input does.
+  return static_cast<double>(cras_client_get_system_max_capture_gain(client_));
+}
+
+void CrasInputStream::SetVolume(double volume) {
+  DCHECK(client_);
+
+  cras_client_set_system_capture_gain(client_, static_cast<long>(volume));
+
+  // Update the AGC volume level based on the last setting above. Note that,
+  // the volume-level resolution is not infinite and it is therefore not
+  // possible to assume that the volume provided as input parameter can be
+  // used directly. Instead, a new query to the audio hardware is required.
+  // This method does nothing if AGC is disabled.
+  UpdateAgcVolume();
+}
+
+double CrasInputStream::GetVolume() {
+  if (!client_)
+    return 0.0;
+
+  long current_volume = cras_client_get_system_capture_gain(client_);
+
+  // Only want the gain, limit reported level to zero.
+  if (current_volume < 0)
+    return 0.0;
+
+  return static_cast<double>(current_volume);
+}
+
+}  // namespace media