| Index: media/audio/pulse/pulse_output.cc
|
| diff --git a/media/audio/pulse/pulse_output.cc b/media/audio/pulse/pulse_output.cc
|
| index 82f9fb926a8ade41747b76db34562c85de3fde20..0687e6ef3b5c4984068208acf46a7fd45e47039d 100644
|
| --- a/media/audio/pulse/pulse_output.cc
|
| +++ b/media/audio/pulse/pulse_output.cc
|
| @@ -4,41 +4,45 @@
|
|
|
| #include "media/audio/pulse/pulse_output.h"
|
|
|
| -#include "base/bind.h"
|
| +#include <pulse/pulseaudio.h>
|
| +
|
| #include "base/message_loop.h"
|
| +#include "media/audio/audio_manager_base.h"
|
| #include "media/audio/audio_parameters.h"
|
| #include "media/audio/audio_util.h"
|
| -#if defined(OS_LINUX)
|
| -#include "media/audio/linux/audio_manager_linux.h"
|
| -#elif defined(OS_OPENBSD)
|
| -#include "media/audio/openbsd/audio_manager_openbsd.h"
|
| -#endif
|
| -#include "media/base/data_buffer.h"
|
| -#include "media/base/seekable_buffer.h"
|
|
|
| namespace media {
|
|
|
| +// A helper class that acquires pa_threaded_mainloop_lock() while in scope.
|
| +class AutoPulseLock {
|
| + public:
|
| + explicit AutoPulseLock(pa_threaded_mainloop* pa_mainloop)
|
| + : pa_mainloop_(pa_mainloop) {
|
| + pa_threaded_mainloop_lock(pa_mainloop_);
|
| + }
|
| +
|
| + ~AutoPulseLock() {
|
| + pa_threaded_mainloop_unlock(pa_mainloop_);
|
| + }
|
| +
|
| + private:
|
| + pa_threaded_mainloop* pa_mainloop_;
|
| +
|
| + DISALLOW_COPY_AND_ASSIGN(AutoPulseLock);
|
| +};
|
| +
|
| static pa_sample_format_t BitsToPASampleFormat(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:
|
| + NOTREACHED() << "Invalid bits per sample: " << bits_per_sample;
|
| return PA_SAMPLE_INVALID;
|
| }
|
| }
|
| @@ -71,202 +75,245 @@ static pa_channel_position ChromiumToPAChannelPosition(Channels channel) {
|
| return PA_CHANNEL_POSITION_SIDE_RIGHT;
|
| case CHANNELS_MAX:
|
| return PA_CHANNEL_POSITION_INVALID;
|
| + default:
|
| + NOTREACHED() << "Invalid channel: " << channel;
|
| + return PA_CHANNEL_POSITION_INVALID;
|
| }
|
| - NOTREACHED() << "Invalid channel " << channel;
|
| - return PA_CHANNEL_POSITION_INVALID;
|
| }
|
|
|
| static pa_channel_map ChannelLayoutToPAChannelMap(
|
| ChannelLayout channel_layout) {
|
| - // Initialize channel map.
|
| pa_channel_map channel_map;
|
| pa_channel_map_init(&channel_map);
|
|
|
| channel_map.channels = ChannelLayoutToChannelCount(channel_layout);
|
| + for (Channels ch = LEFT; ch < CHANNELS_MAX;
|
| + ch = static_cast<Channels>(ch + 1)) {
|
| + int channel_index = ChannelOrder(channel_layout, ch);
|
| + if (channel_index < 0)
|
| + continue;
|
|
|
| - // All channel maps have the same size array of channel positions.
|
| - for (unsigned int channel = 0; channel != CHANNELS_MAX; ++channel) {
|
| - int channel_position = kChannelOrderings[channel_layout][channel];
|
| - if (channel_position > -1) {
|
| - channel_map.map[channel_position] = ChromiumToPAChannelPosition(
|
| - static_cast<Channels>(channel));
|
| - } else {
|
| - // PulseAudio expects unused channels in channel maps to be filled with
|
| - // PA_CHANNEL_POSITION_MONO.
|
| - channel_map.map[channel_position] = PA_CHANNEL_POSITION_MONO;
|
| - }
|
| - }
|
| -
|
| - // Fill in the rest of the unused channels.
|
| - for (unsigned int channel = CHANNELS_MAX; channel != PA_CHANNELS_MAX;
|
| - ++channel) {
|
| - channel_map.map[channel] = PA_CHANNEL_POSITION_MONO;
|
| + channel_map.map[channel_index] = ChromiumToPAChannelPosition(ch);
|
| }
|
|
|
| return channel_map;
|
| }
|
|
|
| -static size_t MicrosecondsToBytes(
|
| - uint32 microseconds, uint32 sample_rate, size_t bytes_per_frame) {
|
| - return microseconds * sample_rate * bytes_per_frame /
|
| - base::Time::kMicrosecondsPerSecond;
|
| -}
|
| +// static, pa_context_notify_cb
|
| +void PulseAudioOutputStream::ContextNotifyCallback(pa_context* c,
|
| + void* p_this) {
|
| + PulseAudioOutputStream* stream = static_cast<PulseAudioOutputStream*>(p_this);
|
| +
|
| + // Forward unexpected failures to the AudioSourceCallback if available. All
|
| + // these variables are only modified under pa_threaded_mainloop_lock() so this
|
| + // should be thread safe.
|
| + if (c && stream->source_callback_ &&
|
| + pa_context_get_state(c) == PA_CONTEXT_FAILED) {
|
| + stream->source_callback_->OnError(stream, pa_context_errno(c));
|
| + }
|
|
|
| -// static
|
| -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);
|
| + pa_threaded_mainloop_signal(stream->pa_mainloop_, 0);
|
| }
|
|
|
| -// static
|
| -void PulseAudioOutputStream::WriteRequestCallback(pa_stream* playback_handle,
|
| - size_t length,
|
| - void* stream_addr) {
|
| - PulseAudioOutputStream* stream =
|
| - reinterpret_cast<PulseAudioOutputStream*>(stream_addr);
|
| +// static, pa_stream_notify_cb
|
| +void PulseAudioOutputStream::StreamNotifyCallback(pa_stream* s, void* p_this) {
|
| + PulseAudioOutputStream* stream = static_cast<PulseAudioOutputStream*>(p_this);
|
| +
|
| + // Forward unexpected failures to the AudioSourceCallback if available. All
|
| + // these variables are only modified under pa_threaded_mainloop_lock() so this
|
| + // should be thread safe.
|
| + if (s && stream->source_callback_ &&
|
| + pa_stream_get_state(s) == PA_STREAM_FAILED) {
|
| + stream->source_callback_->OnError(
|
| + stream, pa_context_errno(stream->pa_context_));
|
| + }
|
|
|
| - DCHECK(stream->manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| + pa_threaded_mainloop_signal(stream->pa_mainloop_, 0);
|
| +}
|
|
|
| - stream->write_callback_handled_ = true;
|
| +// static, pa_stream_success_cb_t
|
| +void PulseAudioOutputStream::StreamSuccessCallback(pa_stream* s, int success,
|
| + void* p_this) {
|
| + PulseAudioOutputStream* stream = static_cast<PulseAudioOutputStream*>(p_this);
|
| + pa_threaded_mainloop_signal(stream->pa_mainloop_, 0);
|
| +}
|
|
|
| - // Fulfill write request.
|
| - stream->FulfillWriteRequest(length);
|
| +// static, pa_stream_request_cb_t
|
| +void PulseAudioOutputStream::StreamRequestCallback(pa_stream* s, size_t len,
|
| + void* p_this) {
|
| + // Fulfill write request; must always result in a pa_stream_write() call.
|
| + static_cast<PulseAudioOutputStream*>(p_this)->FulfillWriteRequest(len);
|
| }
|
|
|
| PulseAudioOutputStream::PulseAudioOutputStream(const AudioParameters& params,
|
| - AudioManagerPulse* manager)
|
| - : channel_layout_(params.channel_layout()),
|
| - channel_count_(ChannelLayoutToChannelCount(channel_layout_)),
|
| - sample_format_(BitsToPASampleFormat(params.bits_per_sample())),
|
| - sample_rate_(params.sample_rate()),
|
| - bytes_per_frame_(params.GetBytesPerFrame()),
|
| + AudioManagerBase* manager)
|
| + : params_(params),
|
| manager_(manager),
|
| pa_context_(NULL),
|
| pa_mainloop_(NULL),
|
| - playback_handle_(NULL),
|
| - packet_size_(params.GetBytesPerBuffer()),
|
| - frames_per_packet_(packet_size_ / bytes_per_frame_),
|
| - client_buffer_(NULL),
|
| + pa_stream_(NULL),
|
| volume_(1.0f),
|
| - stream_stopped_(true),
|
| - write_callback_handled_(false),
|
| - ALLOW_THIS_IN_INITIALIZER_LIST(weak_factory_(this)),
|
| source_callback_(NULL) {
|
| DCHECK(manager_->GetMessageLoop()->BelongsToCurrentThread());
|
|
|
| - // TODO(slock): Sanity check input values.
|
| + CHECK(params_.IsValid());
|
| + audio_bus_ = AudioBus::Create(params_);
|
| }
|
|
|
| PulseAudioOutputStream::~PulseAudioOutputStream() {
|
| - // All internal structures should already have been freed in Close(),
|
| - // which calls AudioManagerPulse::Release which deletes this object.
|
| - DCHECK(!playback_handle_);
|
| + // All internal structures should already have been freed in Close(), which
|
| + // calls AudioManagerBase::ReleaseOutputStream() which deletes this object.
|
| + DCHECK(!pa_stream_);
|
| DCHECK(!pa_context_);
|
| DCHECK(!pa_mainloop_);
|
| }
|
|
|
| +// Helper macro for Open() to avoid code spam and string bloat.
|
| +#define RETURN_ON_FAILURE(expression, message) do { \
|
| + if (!(expression)) { \
|
| + if (pa_context_) { \
|
| + DLOG(ERROR) << message << " Error: " \
|
| + << pa_strerror(pa_context_errno(pa_context_)); \
|
| + } else { \
|
| + DLOG(ERROR) << message; \
|
| + } \
|
| + return false; \
|
| + } \
|
| +} while(0)
|
| +
|
| bool PulseAudioOutputStream::Open() {
|
| DCHECK(manager_->GetMessageLoop()->BelongsToCurrentThread());
|
|
|
| - // TODO(slock): Possibly move most of this to an OpenPlaybackDevice function
|
| - // in a new class 'pulse_util', like alsa_util.
|
| + pa_mainloop_ = pa_threaded_mainloop_new();
|
| + RETURN_ON_FAILURE(pa_mainloop_, "Failed to create PulseAudio main loop.");
|
|
|
| - // 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_mainloop_api* pa_mainloop_api = pa_threaded_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;
|
| - }
|
| + RETURN_ON_FAILURE(pa_context_, "Failed to create PulseAudio context.");
|
| +
|
| + // A state callback must be set before calling pa_threaded_mainloop_lock() or
|
| + // pa_threaded_mainloop_wait() calls may lead to dead lock.
|
| + pa_context_set_state_callback(pa_context_, &ContextNotifyCallback, this);
|
| +
|
| + // Lock the main loop while setting up the context. Failure to do so may lead
|
| + // to crashes as the PulseAudio thread tries to run before things are ready.
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| +
|
| + RETURN_ON_FAILURE(
|
| + pa_threaded_mainloop_start(pa_mainloop_) == 0,
|
| + "Failed to start PulseAudio main loop.");
|
| + RETURN_ON_FAILURE(
|
| + pa_context_connect(pa_context_, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL) == 0,
|
| + "Failed to connect PulseAudio context.");
|
| +
|
| + // Wait until |pa_context_| is ready. pa_threaded_mainloop_wait() must be
|
| + // called after pa_context_get_state() in case the context is already ready,
|
| + // otherwise pa_threaded_mainloop_wait() will hang indefinitely.
|
| + while (true) {
|
| + pa_context_state_t context_state = pa_context_get_state(pa_context_);
|
| + RETURN_ON_FAILURE(
|
| + PA_CONTEXT_IS_GOOD(context_state), "Invalid PulseAudio context state.");
|
| + if (context_state == PA_CONTEXT_READY)
|
| + break;
|
| + pa_threaded_mainloop_wait(pa_mainloop_);
|
| }
|
|
|
| // Set sample specifications.
|
| pa_sample_spec pa_sample_specifications;
|
| - pa_sample_specifications.format = sample_format_;
|
| - pa_sample_specifications.rate = sample_rate_;
|
| - pa_sample_specifications.channels = channel_count_;
|
| + 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 playback stream.
|
| pa_channel_map* map = NULL;
|
| pa_channel_map source_channel_map = ChannelLayoutToPAChannelMap(
|
| - channel_layout_);
|
| + params_.channel_layout());
|
| if (source_channel_map.channels != 0) {
|
| // The source data uses a supported channel map so we will use it rather
|
| // than the default channel map (NULL).
|
| map = &source_channel_map;
|
| }
|
| - playback_handle_ = pa_stream_new(pa_context_, "Playback",
|
| - &pa_sample_specifications, map);
|
| + pa_stream_ = pa_stream_new(
|
| + pa_context_, "Playback", &pa_sample_specifications, map);
|
| + RETURN_ON_FAILURE(pa_stream_, "Failed to create PulseAudio stream.");
|
| + pa_stream_set_state_callback(pa_stream_, &StreamNotifyCallback, this);
|
|
|
| - // Initialize client buffer.
|
| - uint32 output_packet_size = frames_per_packet_ * bytes_per_frame_;
|
| - client_buffer_.reset(new media::SeekableBuffer(0, output_packet_size));
|
| + // Even though we start the stream corked below, PulseAudio will issue one
|
| + // stream request after setup. FulfillWriteRequest() must fulfill the write.
|
| + pa_stream_set_write_callback(pa_stream_, &StreamRequestCallback, this);
|
|
|
| - // Set write callback.
|
| - pa_stream_set_write_callback(playback_handle_, &WriteRequestCallback, this);
|
| -
|
| - // 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.
|
| + // Tell pulse audio we only want callbacks of a certain size.
|
| 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.maxlength = params_.GetBytesPerBuffer();
|
| + pa_buffer_attributes.minreq = params_.GetBytesPerBuffer();
|
| + pa_buffer_attributes.prebuf = params_.GetBytesPerBuffer();
|
| + pa_buffer_attributes.tlength = params_.GetBytesPerBuffer();
|
| 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;
|
| + // TODO(dalecurtis): Pulse tends to want really large buffer sizes if we are
|
| + // not using the native sample rate. We should always open the stream with
|
| + // PA_STREAM_FIX_RATE and ensure this is true.
|
| + RETURN_ON_FAILURE(
|
| + pa_stream_connect_playback(
|
| + pa_stream_, NULL, &pa_buffer_attributes,
|
| + static_cast<pa_stream_flags_t>(
|
| + PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE |
|
| + PA_STREAM_NOT_MONOTONIC | PA_STREAM_START_CORKED),
|
| + NULL, NULL) == 0,
|
| + "Failed to connect PulseAudio stream.");
|
| +
|
| + // Wait for the stream to be ready.
|
| + while (true) {
|
| + pa_stream_state_t stream_state = pa_stream_get_state(pa_stream_);
|
| + RETURN_ON_FAILURE(
|
| + PA_STREAM_IS_GOOD(stream_state), "Invalid PulseAudio stream state.");
|
| + if (stream_state == PA_STREAM_READY)
|
| + break;
|
| + pa_threaded_mainloop_wait(pa_mainloop_);
|
| }
|
|
|
| return true;
|
| }
|
|
|
| +#undef RETURN_ON_FAILURE
|
| +
|
| void PulseAudioOutputStream::Reset() {
|
| - stream_stopped_ = true;
|
| + if (!pa_mainloop_) {
|
| + DCHECK(!pa_stream_);
|
| + DCHECK(!pa_context_);
|
| + return;
|
| + }
|
|
|
| - // Close the stream.
|
| - if (playback_handle_) {
|
| - pa_stream_flush(playback_handle_, NULL, NULL);
|
| - pa_stream_disconnect(playback_handle_);
|
| + {
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| +
|
| + // Close the stream.
|
| + if (pa_stream_) {
|
| + // Ensure all samples are played out before shutdown.
|
| + WaitForPulseOperation(pa_stream_flush(
|
| + pa_stream_, &StreamSuccessCallback, this));
|
| +
|
| + // Release PulseAudio structures.
|
| + pa_stream_disconnect(pa_stream_);
|
| + pa_stream_set_write_callback(pa_stream_, NULL, NULL);
|
| + pa_stream_set_state_callback(pa_stream_, NULL, NULL);
|
| + pa_stream_unref(pa_stream_);
|
| + pa_stream_ = NULL;
|
| + }
|
|
|
| - // 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;
|
| + if (pa_context_) {
|
| + pa_context_disconnect(pa_context_);
|
| + pa_context_set_state_callback(pa_context_, NULL, NULL);
|
| + pa_context_unref(pa_context_);
|
| + pa_context_ = NULL;
|
| + }
|
| }
|
|
|
| - // Release internal buffer.
|
| - client_buffer_.reset();
|
| + pa_threaded_mainloop_stop(pa_mainloop_);
|
| + pa_threaded_mainloop_free(pa_mainloop_);
|
| + pa_mainloop_ = NULL;
|
| }
|
|
|
| void PulseAudioOutputStream::Close() {
|
| @@ -279,138 +326,107 @@ void PulseAudioOutputStream::Close() {
|
| manager_->ReleaseOutputStream(this);
|
| }
|
|
|
| -void PulseAudioOutputStream::WaitForWriteRequest() {
|
| - DCHECK(manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| +int PulseAudioOutputStream::GetHardwareLatencyInBytes() {
|
| + int negative = 0;
|
| + pa_usec_t pa_latency_micros = 0;
|
| + if (pa_stream_get_latency(pa_stream_, &pa_latency_micros, &negative) != 0)
|
| + return 0;
|
|
|
| - if (stream_stopped_)
|
| - return;
|
| + if (negative)
|
| + return 0;
|
|
|
| - // Iterate the PulseAudio mainloop. If 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_) {
|
| - manager_->GetMessageLoop()->PostTask(FROM_HERE, base::Bind(
|
| - &PulseAudioOutputStream::WaitForWriteRequest,
|
| - weak_factory_.GetWeakPtr()));
|
| - }
|
| -}
|
| -
|
| -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_);
|
| - int frames_filled = RunDataCallback(
|
| - audio_bus_.get(), AudioBuffersState(buffer_delay, hardware_delay));
|
| - size_t packet_size = frames_filled * bytes_per_frame_;
|
| -
|
| - DCHECK_LE(packet_size, packet_size_);
|
| - // Note: If this ever changes to output raw float the data must be clipped and
|
| - // sanitized since it may come from an untrusted source such as NaCl.
|
| - audio_bus_->ToInterleaved(
|
| - frames_filled, bytes_per_frame_ / channel_count_,
|
| - packet->GetWritableData());
|
| -
|
| - if (packet_size == 0)
|
| - return false;
|
| -
|
| - 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;
|
| + return (pa_latency_micros * params_.sample_rate() *
|
| + params_.GetBytesPerFrame()) / base::Time::kMicrosecondsPerSecond;
|
| }
|
|
|
| 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;
|
| + CHECK_EQ(requested_bytes, static_cast<size_t>(params_.GetBytesPerBuffer()));
|
|
|
| - // Request more data from the source until we can fulfill the request or
|
| - // fail to receive anymore data.
|
| - bool buffering_successful = true;
|
| - size_t forward_bytes = static_cast<size_t>(client_buffer_->forward_bytes());
|
| - while (forward_bytes < requested_bytes && buffering_successful) {
|
| - buffering_successful = BufferPacketFromSource();
|
| + int frames_filled = 0;
|
| + if (source_callback_) {
|
| + frames_filled = source_callback_->OnMoreData(
|
| + audio_bus_.get(), AudioBuffersState(0, GetHardwareLatencyInBytes()));
|
| }
|
|
|
| - 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
|
| - // the stream as we can.
|
| - WriteToStream(requested_bytes, &bytes_written);
|
| + // Zero any unfilled data so it plays back as silence.
|
| + if (frames_filled < audio_bus_->frames()) {
|
| + audio_bus_->ZeroFramesPartial(
|
| + frames_filled, audio_bus_->frames() - frames_filled);
|
| }
|
|
|
| - if (bytes_written < requested_bytes) {
|
| - // We weren't able to buffer enough data to fulfill the request. Try to
|
| - // fulfill the rest of the request later.
|
| - manager_->GetMessageLoop()->PostTask(FROM_HERE, base::Bind(
|
| - &PulseAudioOutputStream::FulfillWriteRequest,
|
| - weak_factory_.GetWeakPtr(),
|
| - requested_bytes - bytes_written));
|
| - } else {
|
| - // Continue playback.
|
| - manager_->GetMessageLoop()->PostTask(FROM_HERE, base::Bind(
|
| - &PulseAudioOutputStream::WaitForWriteRequest,
|
| - weak_factory_.GetWeakPtr()));
|
| - }
|
| -}
|
| -
|
| -void PulseAudioOutputStream::WriteToStream(size_t bytes_to_write,
|
| - size_t* bytes_written) {
|
| - *bytes_written = 0;
|
| - while (*bytes_written < bytes_to_write) {
|
| - const uint8* chunk;
|
| - int chunk_size;
|
| -
|
| - // Stop writing if there is no more data available.
|
| - if (!client_buffer_->GetCurrentChunk(&chunk, &chunk_size))
|
| - break;
|
| + // PulseAudio won't always be able to provide a buffer large enough, so we may
|
| + // need to request multiple buffers and fill them individually.
|
| + int current_frame = 0;
|
| + size_t bytes_remaining = requested_bytes;
|
| + while (bytes_remaining > 0) {
|
| + void* buffer = NULL;
|
| + size_t bytes_to_fill = bytes_remaining;
|
| + CHECK_GE(pa_stream_begin_write(pa_stream_, &buffer, &bytes_to_fill), 0);
|
| +
|
| + // In case PulseAudio gives us a bigger buffer than we want, cap our size.
|
| + bytes_to_fill = std::min(
|
| + std::min(bytes_remaining, bytes_to_fill),
|
| + static_cast<size_t>(params_.GetBytesPerBuffer()));
|
| +
|
| + int frames_to_fill = bytes_to_fill / params_.GetBytesPerFrame();;
|
| +
|
| + // Note: If this ever changes to output raw float the data must be clipped
|
| + // and sanitized since it may come from an untrusted source such as NaCl.
|
| + audio_bus_->ToInterleavedPartial(
|
| + current_frame, frames_to_fill, params_.bits_per_sample() / 8, buffer);
|
| + media::AdjustVolume(buffer, bytes_to_fill, params_.channels(),
|
| + params_.bits_per_sample() / 8, volume_);
|
| +
|
| + if (pa_stream_write(pa_stream_, buffer, bytes_to_fill, NULL, 0LL,
|
| + PA_SEEK_RELATIVE) < 0) {
|
| + if (source_callback_) {
|
| + source_callback_->OnError(this, pa_context_errno(pa_context_));
|
| + }
|
| + }
|
|
|
| - // 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;
|
| + bytes_remaining -= bytes_to_fill;
|
| + current_frame = frames_to_fill;
|
| }
|
| }
|
|
|
| void PulseAudioOutputStream::Start(AudioSourceCallback* callback) {
|
| DCHECK(manager_->GetMessageLoop()->BelongsToCurrentThread());
|
| CHECK(callback);
|
| - DLOG_IF(ERROR, !playback_handle_)
|
| - << "Open() has not been called successfully";
|
| - if (!playback_handle_)
|
| + CHECK(pa_stream_);
|
| +
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| +
|
| + // Ensure the context and stream are ready.
|
| + if (pa_context_get_state(pa_context_) != PA_CONTEXT_READY &&
|
| + pa_stream_get_state(pa_stream_) != PA_STREAM_READY) {
|
| + callback->OnError(this, pa_context_errno(pa_context_));
|
| return;
|
| + }
|
|
|
| source_callback_ = callback;
|
|
|
| - // Clear buffer, it might still have data in it.
|
| - client_buffer_->Clear();
|
| - stream_stopped_ = false;
|
| -
|
| - // Start playback.
|
| - manager_->GetMessageLoop()->PostTask(FROM_HERE, base::Bind(
|
| - &PulseAudioOutputStream::WaitForWriteRequest,
|
| - weak_factory_.GetWeakPtr()));
|
| + // Uncork (resume) the stream.
|
| + WaitForPulseOperation(pa_stream_cork(
|
| + pa_stream_, 0, &StreamSuccessCallback, this));
|
| }
|
|
|
| void PulseAudioOutputStream::Stop() {
|
| DCHECK(manager_->GetMessageLoop()->BelongsToCurrentThread());
|
|
|
| - stream_stopped_ = true;
|
| + // Cork (pause) the stream. Waiting for the main loop lock will ensure
|
| + // outstanding callbacks have completed.
|
| + AutoPulseLock auto_lock(pa_mainloop_);
|
| +
|
| + // Flush the stream prior to cork, doing so after will cause hangs. Write
|
| + // callbacks are suspended while inside pa_threaded_mainloop_lock() so this
|
| + // is all thread safe.
|
| + WaitForPulseOperation(pa_stream_flush(
|
| + pa_stream_, &StreamSuccessCallback, this));
|
| +
|
| + WaitForPulseOperation(pa_stream_cork(
|
| + pa_stream_, 1, &StreamSuccessCallback, this));
|
| +
|
| + source_callback_ = NULL;
|
| }
|
|
|
| void PulseAudioOutputStream::SetVolume(double volume) {
|
| @@ -425,12 +441,12 @@ void PulseAudioOutputStream::GetVolume(double* volume) {
|
| *volume = volume_;
|
| }
|
|
|
| -int PulseAudioOutputStream::RunDataCallback(
|
| - AudioBus* audio_bus, AudioBuffersState buffers_state) {
|
| - if (source_callback_)
|
| - return source_callback_->OnMoreData(audio_bus, buffers_state);
|
| -
|
| - return 0;
|
| +void PulseAudioOutputStream::WaitForPulseOperation(pa_operation* op) {
|
| + CHECK(op);
|
| + while (pa_operation_get_state(op) == PA_OPERATION_RUNNING) {
|
| + pa_threaded_mainloop_wait(pa_mainloop_);
|
| + }
|
| + pa_operation_unref(op);
|
| }
|
|
|
| } // namespace media
|
|
|