blob: 74a0b32ec11952d3a4b4c09615edc6a1be540b01 [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 "ppapi/shared_impl/ppb_audio_config_shared.h"
#include <algorithm>
#include "build/build_config.h"
#include "ppapi/thunk/enter.h"
#include "ppapi/thunk/ppb_instance_api.h"
namespace ppapi {
// Rounds up requested_size to the nearest multiple of minimum_size.
static uint32_t CalculateMultipleOfSampleFrameCount(uint32_t minimum_size,
uint32_t requested_size) {
const uint32_t multiple = (requested_size + minimum_size - 1) / minimum_size;
return std::min(minimum_size * multiple,
PPB_AudioConfig_Shared::PPB_AudioConfig_Shared(ResourceObjectType type,
PP_Instance instance)
: Resource(type, instance),
sample_frame_count_(0) {}
PPB_AudioConfig_Shared::~PPB_AudioConfig_Shared() {}
PP_Resource PPB_AudioConfig_Shared::Create(ResourceObjectType type,
PP_Instance instance,
PP_AudioSampleRate sample_rate,
uint32_t sample_frame_count) {
scoped_refptr<PPB_AudioConfig_Shared> object(
new PPB_AudioConfig_Shared(type, instance));
if (!object->Init(sample_rate, sample_frame_count))
return 0;
return object->GetReference();
// static
uint32_t PPB_AudioConfig_Shared::RecommendSampleFrameCount_1_0(
PP_AudioSampleRate sample_rate,
uint32_t requested_sample_frame_count) {
// Version 1.0: Don't actually query to get a value from the
// hardware; instead return the input for in-range values.
if (requested_sample_frame_count < PP_AUDIOMINSAMPLEFRAMECOUNT)
if (requested_sample_frame_count > PP_AUDIOMAXSAMPLEFRAMECOUNT)
return requested_sample_frame_count;
// static
uint32_t PPB_AudioConfig_Shared::RecommendSampleFrameCount_1_1(
PP_Instance instance,
PP_AudioSampleRate sample_rate,
uint32_t sample_frame_count) {
// Version 1.1: Query the back-end hardware for sample rate and buffer size,
// and recommend a best fit based on request.
thunk::EnterInstanceNoLock enter(instance);
if (enter.failed())
return 0;
// Get the hardware config.
PP_AudioSampleRate hardware_sample_rate = static_cast<PP_AudioSampleRate>(
uint32_t hardware_sample_frame_count =
if (sample_frame_count < PP_AUDIOMINSAMPLEFRAMECOUNT)
// If hardware information isn't available we're connected to a fake audio
// output stream on the browser side, so we can use whatever sample count the
// client wants.
if (!hardware_sample_frame_count || !hardware_sample_rate)
return sample_frame_count;
// Note: All the values below were determined through experimentation to
// minimize jitter and back-to-back callbacks from the browser. Please take
// care when modifying these values as they impact a large number of users.
// TODO(dalecurtis): Land jitter test and add documentation for updating this.
// Should track the value reported by XP and ALSA backends.
const uint32_t kHighLatencySampleFrameCount = 2048;
#if defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
// TODO(ihf): Remove this once ARM Chromebooks support low latency audio. For
// now we classify them as high latency. See Note that
// Adobe Flash is affected but not HTML5, WebRTC and WebAudio (they are using
// real time threads).
const bool kHighLatencyDevice = true;
const bool kHighLatencyDevice = false;
// If client is using same sample rate as audio hardware, then recommend a
// multiple of the audio hardware's sample frame count.
if (!kHighLatencyDevice && hardware_sample_rate == sample_rate) {
return CalculateMultipleOfSampleFrameCount(hardware_sample_frame_count,
// If the hardware requires a high latency buffer or we're at a low sample
// rate w/ a buffer that's larger than 10ms, choose the nearest multiple of
// the high latency sample frame count. An example of too low and too large
// is 16kHz and a sample frame count greater than 160 frames.
if (kHighLatencyDevice ||
hardware_sample_frame_count >= kHighLatencySampleFrameCount ||
(hardware_sample_rate < 44100 &&
hardware_sample_frame_count > hardware_sample_rate / 100u)) {
return CalculateMultipleOfSampleFrameCount(
std::max(kHighLatencySampleFrameCount, hardware_sample_frame_count));
// All low latency clients should be able to handle a 512 frame buffer with
// resampling from 44.1kHz and 48kHz to higher sample rates.
// TODO(dalecurtis): We may need to investigate making the callback thread
// high priority to handle buffers at the absolute minimum w/o glitching.
const uint32_t kLowLatencySampleFrameCount = 512;
// Special case for 48kHz -> 44.1kHz and buffer sizes greater than 10ms. In
// testing most buffer sizes > 10ms led to glitching, so we choose a size we
// know won't cause jitter.
int min_sample_frame_count = kLowLatencySampleFrameCount;
if (hardware_sample_rate == 44100 && sample_rate == 48000 &&
hardware_sample_frame_count > hardware_sample_rate / 100u) {
min_sample_frame_count =
std::max(2 * kLowLatencySampleFrameCount, hardware_sample_frame_count);
return CalculateMultipleOfSampleFrameCount(min_sample_frame_count,
// static
PP_AudioSampleRate PPB_AudioConfig_Shared::RecommendSampleRate(
PP_Instance instance) {
thunk::EnterInstanceNoLock enter(instance);
if (enter.failed())
PP_AudioSampleRate hardware_sample_rate = static_cast<PP_AudioSampleRate>(
return hardware_sample_rate;
thunk::PPB_AudioConfig_API* PPB_AudioConfig_Shared::AsPPB_AudioConfig_API() {
return this;
PP_AudioSampleRate PPB_AudioConfig_Shared::GetSampleRate() {
return sample_rate_;
uint32_t PPB_AudioConfig_Shared::GetSampleFrameCount() {
return sample_frame_count_;
bool PPB_AudioConfig_Shared::Init(PP_AudioSampleRate sample_rate,
uint32_t sample_frame_count) {
// TODO(brettw): Currently we don't actually check what the hardware
// supports, so just allow sample rates of the "guaranteed working" ones.
// TODO(dalecurtis): If sample rates are added RecommendSampleFrameCount_1_1()
// must be updated to account for the new rates.
if (sample_rate != PP_AUDIOSAMPLERATE_44100 &&
sample_rate != PP_AUDIOSAMPLERATE_48000)
return false;
// TODO(brettw): Currently we don't actually query to get a value from the
// hardware, so just validate the range.
if (sample_frame_count > PP_AUDIOMAXSAMPLEFRAMECOUNT ||
return false;
sample_rate_ = sample_rate;
sample_frame_count_ = sample_frame_count;
return true;
} // namespace ppapi