| Index: media/audio/pulse/pulse_input.cc
|
| diff --git a/media/audio/pulse/pulse_input.cc b/media/audio/pulse/pulse_input.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..d70dc3e8a18771140e4a15e03384a2d5b71e1605
|
| --- /dev/null
|
| +++ b/media/audio/pulse/pulse_input.cc
|
| @@ -0,0 +1,320 @@
|
| +// 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/pulse/pulse_input.h"
|
| +
|
| +#include "base/logging.h"
|
| +#include "base/message_loop.h"
|
| +#include "media/audio/audio_manager_base.h"
|
| +#include "media/audio/pulse/pulse_util.h"
|
| +#include "media/base/seekable_buffer.h"
|
| +
|
| +namespace media {
|
| +
|
| +PulseAudioInputStream::PulseAudioInputStream(AudioManagerBase* audio_manager,
|
| + const std::string& device_name,
|
| + const AudioParameters& params,
|
| + pa_threaded_mainloop* mainloop,
|
| + pa_context* context)
|
| + : audio_manager_(audio_manager),
|
| + callback_(NULL),
|
| + device_name_(device_name),
|
| + params_(params),
|
| + channels_(0),
|
| + volume_(0.0),
|
| + stream_started_(false),
|
| + pa_mainloop_(mainloop),
|
| + pa_context_(context),
|
| + handle_(NULL) {
|
| + DCHECK(audio_manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| + DCHECK(mainloop);
|
| + DCHECK(context);
|
| +}
|
| +
|
| +PulseAudioInputStream::~PulseAudioInputStream() {
|
| + // All internal structures should already have been freed in Close(),
|
| + // which calls AudioManagerPulse::Release which deletes this object.
|
| + DCHECK(!handle_);
|
| +}
|
| +
|
| +bool PulseAudioInputStream::Open() {
|
| + DCHECK(audio_manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| +
|
| + // Set sample specifications.
|
| + pa_sample_spec pa_sample_specifications;
|
| + pa_sample_specifications.format = BitsToPASampleFormat(
|
| + params_.bits_per_sample());
|
| + pa_sample_specifications.rate = params_.sample_rate();
|
| + pa_sample_specifications.channels = params_.channels();
|
| +
|
| + // Get channel mapping and open recording stream.
|
| + pa_channel_map source_channel_map = ChannelLayoutToPAChannelMap(
|
| + params_.channel_layout());
|
| + pa_channel_map* map = (source_channel_map.channels != 0)?
|
| + &source_channel_map : NULL;
|
| +
|
| + // Create a new recording stream.
|
| + handle_ = pa_stream_new(pa_context_, "RecordStream",
|
| + &pa_sample_specifications, map);
|
| + if (!handle_) {
|
| + DLOG(ERROR) << "Open: failed to create PA stream";
|
| + return false;
|
| + }
|
| +
|
| + pa_stream_set_state_callback(handle_, &StreamNotifyCallback, this);
|
| + pa_stream_set_read_callback(handle_, &ReadCallback, this);
|
| + pa_stream_readable_size(handle_);
|
| +
|
| + // Set server-side capture buffer metrics. Detailed documentation on what
|
| + // values should be chosen can be found at
|
| + // freedesktop.org/software/pulseaudio/doxygen/structpa__buffer__attr.html.
|
| + pa_buffer_attr buffer_attributes;
|
| + const unsigned int buffer_size = params_.GetBytesPerBuffer();
|
| + buffer_attributes.maxlength = static_cast<uint32_t>(-1);
|
| + buffer_attributes.tlength = buffer_size;
|
| + buffer_attributes.minreq = buffer_size;
|
| + buffer_attributes.prebuf = static_cast<uint32_t>(-1);
|
| + buffer_attributes.fragsize = buffer_size;
|
| + int flags = PA_STREAM_AUTO_TIMING_UPDATE |
|
| + PA_STREAM_INTERPOLATE_TIMING |
|
| + PA_STREAM_ADJUST_LATENCY |
|
| + PA_STREAM_START_CORKED;
|
| + int err = pa_stream_connect_record(
|
| + handle_,
|
| + device_name_ == AudioManagerBase::kDefaultDeviceId ?
|
| + NULL : device_name_.c_str(),
|
| + &buffer_attributes,
|
| + static_cast<pa_stream_flags_t>(flags));
|
| + if (err) {
|
| + DLOG(ERROR) << "pa_stream_connect_playback FAILED " << err;
|
| + return false;
|
| + }
|
| +
|
| + // Wait for the stream to be ready.
|
| + while (true) {
|
| + pa_stream_state_t stream_state = pa_stream_get_state(handle_);
|
| + if(!PA_STREAM_IS_GOOD(stream_state)) {
|
| + DLOG(ERROR) << "Invalid PulseAudio stream state";
|
| + return false;
|
| + }
|
| +
|
| + if (stream_state == PA_STREAM_READY)
|
| + break;
|
| + pa_threaded_mainloop_wait(pa_mainloop_);
|
| + }
|
| +
|
| + pa_stream_set_read_callback(handle_, &ReadCallback, this);
|
| + pa_stream_readable_size(handle_);
|
| +
|
| + buffer_.reset(new media::SeekableBuffer(0, 2 * params_.GetBytesPerBuffer()));
|
| + audio_data_buffer_.reset(new uint8[params_.GetBytesPerBuffer()]);
|
| + return true;
|
| +}
|
| +
|
| +void PulseAudioInputStream::Start(AudioInputCallback* callback) {
|
| + DCHECK(audio_manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| + DCHECK(callback);
|
| + DCHECK(handle_);
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| +
|
| + if (stream_started_)
|
| + return;
|
| +
|
| + // Clean up the old buffer.
|
| + pa_stream_drop(handle_);
|
| +
|
| + // Start the streaming.
|
| + stream_started_ = true;
|
| + callback_ = callback;
|
| +
|
| + pa_operation* operation = pa_stream_cork(handle_, 0, NULL, NULL);
|
| + WaitForOperationCompletion(pa_mainloop_, operation);
|
| +}
|
| +
|
| +void PulseAudioInputStream::Stop() {
|
| + DCHECK(audio_manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| + if (!stream_started_)
|
| + return;
|
| +
|
| + // Set the flag to false to stop filling new data to soundcard.
|
| + stream_started_ = false;
|
| +
|
| + pa_operation* operation = pa_stream_flush(
|
| + handle_, &StreamSuccessCallback, pa_mainloop_);
|
| + WaitForOperationCompletion(pa_mainloop_, operation);
|
| +
|
| + // Stop the stream.
|
| + pa_stream_set_read_callback(handle_, NULL, NULL);
|
| + operation = pa_stream_cork(handle_, 1, &StreamSuccessCallback, pa_mainloop_);
|
| + WaitForOperationCompletion(pa_mainloop_, operation);
|
| +}
|
| +
|
| +void PulseAudioInputStream::Close() {
|
| + DCHECK(audio_manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| + {
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| + if (handle_) {
|
| + // Disable all the callbacks before disconnecting.
|
| + pa_stream_set_state_callback(handle_, NULL, NULL);
|
| + pa_stream_flush(handle_, NULL, NULL);
|
| +
|
| + if (pa_stream_get_state(handle_) != PA_STREAM_UNCONNECTED)
|
| + pa_stream_disconnect(handle_);
|
| +
|
| + // Release PulseAudio structures.
|
| + pa_stream_unref(handle_);
|
| + handle_ = NULL;
|
| + }
|
| + }
|
| +
|
| + if (callback_)
|
| + callback_->OnClose(this);
|
| +
|
| + // 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".
|
| + audio_manager_->ReleaseInputStream(this);
|
| +}
|
| +
|
| +double PulseAudioInputStream::GetMaxVolume() {
|
| + return static_cast<double>(PA_VOLUME_NORM);
|
| +}
|
| +
|
| +void PulseAudioInputStream::SetVolume(double volume) {
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| + if (!handle_)
|
| + return;
|
| +
|
| + size_t index = pa_stream_get_device_index(handle_);
|
| + pa_operation* operation = NULL;
|
| + if (!channels_) {
|
| + // Get the number of channels for the source only when the |channels_| is 0.
|
| + // We are assuming the stream source is not changed on the fly here.
|
| + operation = pa_context_get_source_info_by_index(
|
| + pa_context_, index, &VolumeCallback, this);
|
| + WaitForOperationCompletion(pa_mainloop_, operation);
|
| + if (!channels_) {
|
| + DLOG(WARNING) << "Failed to get the number of channels for the source";
|
| + return;
|
| + }
|
| + }
|
| +
|
| + pa_cvolume pa_volume;
|
| + pa_cvolume_set(&pa_volume, channels_, volume);
|
| + operation = pa_context_set_source_volume_by_index(
|
| + pa_context_, index, &pa_volume, NULL, NULL);
|
| +
|
| + // Don't need to wait for this task to complete.
|
| + pa_operation_unref(operation);
|
| +}
|
| +
|
| +double PulseAudioInputStream::GetVolume() {
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| + if (!handle_)
|
| + return 0.0;
|
| +
|
| + size_t index = pa_stream_get_device_index(handle_);
|
| + pa_operation* operation = pa_context_get_source_info_by_index(
|
| + pa_context_, index, &VolumeCallback, this);
|
| + WaitForOperationCompletion(pa_mainloop_, operation);
|
| +
|
| + return volume_;
|
| +}
|
| +
|
| +// static, used by pa_stream_set_read_callback.
|
| +void PulseAudioInputStream::ReadCallback(pa_stream* handle,
|
| + size_t length,
|
| + void* user_data) {
|
| + PulseAudioInputStream* stream =
|
| + reinterpret_cast<PulseAudioInputStream*>(user_data);
|
| +
|
| + stream->ReadData();
|
| +}
|
| +
|
| +// static, used by pa_context_get_source_info_by_index.
|
| +void PulseAudioInputStream::VolumeCallback(pa_context* context,
|
| + const pa_source_info* info,
|
| + int error, void* user_data) {
|
| + PulseAudioInputStream* stream =
|
| + reinterpret_cast<PulseAudioInputStream*>(user_data);
|
| +
|
| + if (error) {
|
| + pa_threaded_mainloop_signal(stream->pa_mainloop_, 0);
|
| + return;
|
| + }
|
| +
|
| + if (stream->channels_ != info->channel_map.channels)
|
| + stream->channels_ = info->channel_map.channels;
|
| +
|
| + pa_volume_t volume = PA_VOLUME_MUTED; // Minimum possible value.
|
| + // Use the max volume of any channel as the volume.
|
| + for (int i = 0; i < stream->channels_; ++i) {
|
| + if (volume < info->volume.values[i])
|
| + volume = info->volume.values[i];
|
| + }
|
| +
|
| + stream->volume_ = static_cast<double>(volume);
|
| +}
|
| +
|
| +// static, used by pa_stream_set_state_callback.
|
| +void PulseAudioInputStream::StreamNotifyCallback(pa_stream* stream,
|
| + void* user_data) {
|
| + PulseAudioInputStream* pulse_stream =
|
| + reinterpret_cast<PulseAudioInputStream*>(user_data);
|
| + if (stream && pulse_stream->callback_ &&
|
| + pa_stream_get_state(stream) == PA_STREAM_FAILED) {
|
| + pulse_stream->callback_->OnError(
|
| + pulse_stream, pa_context_errno(pulse_stream->pa_context_));
|
| + }
|
| +
|
| + pa_threaded_mainloop_signal(pulse_stream->pa_mainloop_, 0);
|
| +}
|
| +
|
| +void PulseAudioInputStream::ReadData() {
|
| + uint32 hardware_delay = GetHardwareLatencyInBytes(
|
| + handle_, params_.sample_rate(), params_.GetBytesPerFrame());
|
| +
|
| + // 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);
|
| +
|
| + while (true) {
|
| + size_t length = 0;
|
| + const void* data = NULL;
|
| + pa_stream_peek(handle_, &data, &length);
|
| + if (!data || length == 0)
|
| + break;
|
| +
|
| + buffer_->Append(reinterpret_cast<const uint8*>(data), length);
|
| +
|
| + // Checks if we still have data.
|
| + pa_stream_drop(handle_);
|
| + if (pa_stream_readable_size(handle_) <= 0)
|
| + break;
|
| + }
|
| +
|
| + int packet_size = params_.GetBytesPerBuffer();
|
| + while (buffer_->forward_bytes() >= packet_size) {
|
| + buffer_->Read(audio_data_buffer_.get(), packet_size);
|
| + callback_->OnData(this, audio_data_buffer_.get(), packet_size,
|
| + hardware_delay, normalized_volume);
|
| +
|
| + if (buffer_->forward_bytes() < packet_size)
|
| + break;
|
| +
|
| + // TODO(xians): improve the code by implementing a WaitTillDataReady on the
|
| + // input side.
|
| + DLOG(WARNING) << "OnData is being called consecutively, sleep 2ms to "
|
| + << "wait until render consumes the data";
|
| + base::PlatformThread::Sleep(
|
| + base::TimeDelta::FromMilliseconds(2));
|
| + }
|
| +
|
| + pa_threaded_mainloop_signal(pa_mainloop_, 0);
|
| +}
|
| +
|
| +} // namespace media
|
|
|