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// 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 "base/strings/stringprintf.h"
#include "content/browser/renderer_host/media/media_stream_manager.h"
#include "content/public/common/content_switches.h"
#include "media/audio/audio_parameters.h"
using media::AudioBus;
namespace {
// Used to log if any audio glitches have been detected during an audio session.
// Elements in this enum should not be added, deleted or rearranged.
enum AudioGlitchResult {
AUDIO_RENDERER_NO_AUDIO_GLITCHES = 0,
AUDIO_RENDERER_AUDIO_GLITCHES = 1,
AUDIO_RENDERER_AUDIO_GLITCHES_MAX = AUDIO_RENDERER_AUDIO_GLITCHES
};
void LogAudioGlitchResult(AudioGlitchResult result) {
UMA_HISTOGRAM_ENUMERATION("Media.AudioRendererAudioGlitches",
result,
AUDIO_RENDERER_AUDIO_GLITCHES_MAX + 1);
}
} // namespace
namespace content {
AudioSyncReader::AudioSyncReader(base::SharedMemory* shared_memory,
const media::AudioParameters& params)
: shared_memory_(shared_memory),
mute_audio_(base::CommandLine::ForCurrentProcess()->HasSwitch(
switches::kMuteAudio)),
packet_size_(shared_memory_->requested_size()),
renderer_callback_count_(0),
renderer_missed_callback_count_(0),
#if defined(OS_MACOSX)
maximum_wait_time_(params.GetBufferDuration() / 2),
#else
// TODO(dalecurtis): Investigate if we can reduce this on all platforms.
maximum_wait_time_(base::TimeDelta::FromMilliseconds(20)),
#endif
buffer_index_(0) {
DCHECK_EQ(packet_size_, AudioBus::CalculateMemorySize(params));
output_bus_ = AudioBus::WrapMemory(params, shared_memory->memory());
output_bus_->Zero();
}
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);
// Add more detailed information regarding detected audio glitches where
// a non-zero value of |renderer_missed_callback_count_| is added to the
// AUDIO_RENDERER_AUDIO_GLITCHES bin.
renderer_missed_callback_count_ > 0 ?
LogAudioGlitchResult(AUDIO_RENDERER_AUDIO_GLITCHES) :
LogAudioGlitchResult(AUDIO_RENDERER_NO_AUDIO_GLITCHES);
std::string log_string =
base::StringPrintf("ASR: number of detected audio glitches=%d",
static_cast<int>(renderer_missed_callback_count_));
MediaStreamManager::SendMessageToNativeLog(log_string);
DVLOG(1) << log_string;
}
// media::AudioOutputController::SyncReader implementations.
void AudioSyncReader::UpdatePendingBytes(uint32 bytes) {
// 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();
socket_->Send(&bytes, sizeof(bytes));
++buffer_index_;
}
void AudioSyncReader::Read(AudioBus* dest) {
++renderer_callback_count_;
if (!WaitUntilDataIsReady()) {
++renderer_missed_callback_count_;
dest->Zero();
return;
}
if (mute_audio_)
dest->Zero();
else
output_bus_->CopyTo(dest);
}
void AudioSyncReader::Close() {
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());
}
bool AudioSyncReader::PrepareForeignSocket(
base::ProcessHandle process_handle,
base::SyncSocket::TransitDescriptor* descriptor) {
return foreign_socket_->PrepareTransitDescriptor(process_handle, descriptor);
}
bool AudioSyncReader::WaitUntilDataIsReady() {
base::TimeDelta timeout = maximum_wait_time_;
const base::TimeTicks start_time = base::TimeTicks::Now();
const base::TimeTicks finish_time = start_time + timeout;
// Check if data is ready and if not, wait a reasonable amount of time for it.
//
// Data readiness is achieved via parallel counters, one on the renderer side
// and one here. Every time a buffer is requested via UpdatePendingBytes(),
// |buffer_index_| is incremented. Subsequently every time the renderer has a
// buffer ready it increments its counter and sends the counter value over the
// SyncSocket. Data is ready when |buffer_index_| matches the counter value
// received from the renderer.
//
// The counter values may temporarily become out of sync if the renderer is
// unable to deliver audio fast enough. It's assumed that the renderer will
// catch up at some point, which means discarding counter values read from the
// SyncSocket which don't match our current buffer index.
size_t bytes_received = 0;
uint32 renderer_buffer_index = 0;
while (timeout.InMicroseconds() > 0) {
bytes_received = socket_->ReceiveWithTimeout(
&renderer_buffer_index, sizeof(renderer_buffer_index), timeout);
if (bytes_received != sizeof(renderer_buffer_index)) {
bytes_received = 0;
break;
}
if (renderer_buffer_index == buffer_index_)
break;
// Reduce the timeout value as receives succeed, but aren't the right index.
timeout = finish_time - base::TimeTicks::Now();
}
// Receive timed out or another error occurred. Receive can timeout if the
// renderer is unable to deliver audio data within the allotted time.
if (!bytes_received || renderer_buffer_index != buffer_index_) {
DVLOG(2) << "AudioSyncReader::WaitUntilDataIsReady() timed out.";
base::TimeDelta time_since_start = base::TimeTicks::Now() - start_time;
UMA_HISTOGRAM_CUSTOM_TIMES("Media.AudioOutputControllerDataNotReady",
time_since_start,
base::TimeDelta::FromMilliseconds(1),
base::TimeDelta::FromMilliseconds(1000),
50);
return false;
}
return true;
}
} // namespace content
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