blob: dea8ae2a2075a8b5f7cbfcd0efca8bad86dd1672 [file] [log] [blame]
// 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 "content/browser/renderer_host/media/audio_sync_reader.h"
#include <algorithm>
#include "base/command_line.h"
#include "base/memory/shared_memory.h"
#include "base/metrics/histogram.h"
#include "content/public/common/content_switches.h"
#include "media/audio/audio_buffers_state.h"
#include "media/audio/audio_parameters.h"
#include "media/audio/shared_memory_util.h"
using media::AudioBus;
namespace content {
AudioSyncReader::AudioSyncReader(base::SharedMemory* shared_memory,
const media::AudioParameters& params,
int input_channels)
: shared_memory_(shared_memory),
input_channels_(input_channels),
mute_audio_(CommandLine::ForCurrentProcess()->HasSwitch(
switches::kMuteAudio)),
renderer_callback_count_(0),
renderer_missed_callback_count_(0) {
packet_size_ = media::PacketSizeInBytes(shared_memory_->requested_size());
int input_memory_size = 0;
int output_memory_size = AudioBus::CalculateMemorySize(params);
if (input_channels_ > 0) {
// The input storage is after the output storage.
int frames = params.frames_per_buffer();
input_memory_size = AudioBus::CalculateMemorySize(input_channels_, frames);
char* input_data =
static_cast<char*>(shared_memory_->memory()) + output_memory_size;
input_bus_ = AudioBus::WrapMemory(input_channels_, frames, input_data);
}
DCHECK_EQ(packet_size_, output_memory_size + input_memory_size);
output_bus_ = AudioBus::WrapMemory(params, shared_memory->memory());
}
AudioSyncReader::~AudioSyncReader() {
if (!renderer_callback_count_)
return;
// Recording the percentage of deadline misses gives us a rough overview of
// how many users might be running into audio glitches.
int percentage_missed =
100.0 * renderer_missed_callback_count_ / renderer_callback_count_;
UMA_HISTOGRAM_PERCENTAGE(
"Media.AudioRendererMissedDeadline", percentage_missed);
}
bool AudioSyncReader::DataReady() {
return !media::IsUnknownDataSize(shared_memory_, packet_size_);
}
// media::AudioOutputController::SyncReader implementations.
void AudioSyncReader::UpdatePendingBytes(uint32 bytes) {
if (bytes != static_cast<uint32>(media::kPauseMark)) {
// Store unknown length of data into buffer, so we later
// can find out if data became available.
media::SetUnknownDataSize(shared_memory_, packet_size_);
}
if (socket_) {
socket_->Send(&bytes, sizeof(bytes));
}
}
int AudioSyncReader::Read(bool block, const AudioBus* source, AudioBus* dest) {
++renderer_callback_count_;
if (!DataReady()) {
++renderer_missed_callback_count_;
if (block)
WaitTillDataReady();
}
// Copy optional synchronized live audio input for consumption by renderer
// process.
if (source && input_bus_) {
DCHECK_EQ(source->channels(), input_bus_->channels());
// TODO(crogers): In some cases with device and sample-rate changes
// it's possible for an AOR to insert a resampler in the path.
// Because this is used with the Web Audio API, it'd be better
// to bypass the device change handling in AOR and instead let
// the renderer-side Web Audio code deal with this.
if (source->frames() == input_bus_->frames() &&
source->channels() == input_bus_->channels())
source->CopyTo(input_bus_.get());
else
input_bus_->Zero();
}
// Retrieve the actual number of bytes available from the shared memory. If
// the renderer has not completed rendering this value will be invalid (still
// the marker stored in UpdatePendingBytes() above) and must be sanitized.
// TODO(dalecurtis): Technically this is not the exact size. Due to channel
// padding for alignment, there may be more data available than this; AudioBus
// will automatically do the right thing during CopyTo(). Rename this method
// to GetActualFrameCount().
uint32 size = media::GetActualDataSizeInBytes(shared_memory_, packet_size_);
// Compute the actual number of frames read. It's important to sanitize this
// value for a couple reasons. One, it might still be the unknown data size
// marker. Two, shared memory comes from a potentially untrusted source.
int frames =
size / (sizeof(*output_bus_->channel(0)) * output_bus_->channels());
if (frames < 0)
frames = 0;
else if (frames > output_bus_->frames())
frames = output_bus_->frames();
if (mute_audio_) {
dest->Zero();
} else {
// Copy data from the shared memory into the caller's AudioBus.
output_bus_->CopyTo(dest);
// Zero out any unfilled frames in the destination bus.
dest->ZeroFramesPartial(frames, dest->frames() - frames);
}
// Zero out the entire output buffer to avoid stuttering/repeating-buffers
// in the anomalous case if the renderer is unable to keep up with real-time.
output_bus_->Zero();
// Store unknown length of data into buffer, in case renderer does not store
// the length itself. It also helps in decision if we need to yield.
media::SetUnknownDataSize(shared_memory_, packet_size_);
// Return the actual number of frames read.
return frames;
}
void AudioSyncReader::Close() {
if (socket_) {
socket_->Close();
}
}
bool AudioSyncReader::Init() {
socket_.reset(new base::CancelableSyncSocket());
foreign_socket_.reset(new base::CancelableSyncSocket());
return base::CancelableSyncSocket::CreatePair(socket_.get(),
foreign_socket_.get());
}
#if defined(OS_WIN)
bool AudioSyncReader::PrepareForeignSocketHandle(
base::ProcessHandle process_handle,
base::SyncSocket::Handle* foreign_handle) {
::DuplicateHandle(GetCurrentProcess(), foreign_socket_->handle(),
process_handle, foreign_handle,
0, FALSE, DUPLICATE_SAME_ACCESS);
if (*foreign_handle != 0)
return true;
return false;
}
#else
bool AudioSyncReader::PrepareForeignSocketHandle(
base::ProcessHandle process_handle,
base::FileDescriptor* foreign_handle) {
foreign_handle->fd = foreign_socket_->handle();
foreign_handle->auto_close = false;
if (foreign_handle->fd != -1)
return true;
return false;
}
#endif
void AudioSyncReader::WaitTillDataReady() {
base::TimeTicks start = base::TimeTicks::Now();
const base::TimeDelta kMaxWait = base::TimeDelta::FromMilliseconds(20);
#if defined(OS_WIN)
// Sleep(0) on Windows lets the other threads run.
const base::TimeDelta kSleep = base::TimeDelta::FromMilliseconds(0);
#else
// We want to sleep for a bit here, as otherwise a backgrounded renderer won't
// get enough cpu to send the data and the high priority thread in the browser
// will use up a core causing even more skips.
const base::TimeDelta kSleep = base::TimeDelta::FromMilliseconds(2);
#endif
base::TimeDelta time_since_start;
do {
base::PlatformThread::Sleep(kSleep);
time_since_start = base::TimeTicks::Now() - start;
} while (!DataReady() && time_since_start < kMaxWait);
UMA_HISTOGRAM_CUSTOM_TIMES("Media.AudioOutputControllerDataNotReady",
time_since_start,
base::TimeDelta::FromMilliseconds(1),
base::TimeDelta::FromMilliseconds(1000),
50);
}
} // namespace content