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Kill mutex and locks in osx_render. Improve performance. Also fix an initialization problem

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This commit is contained in:
Yue Wang 2016-09-18 19:40:04 -07:00 committed by GitHub
parent e4b9d679fb
commit 606d029ed9
1 changed files with 91 additions and 131 deletions
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222
src/output/plugins/OSXOutputPlugin.cxx
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@ -30,7 +30,7 @@
#include <CoreAudio/CoreAudio.h> #include <CoreAudio/CoreAudio.h>
#include <AudioUnit/AudioUnit.h> #include <AudioUnit/AudioUnit.h>
#include <CoreServices/CoreServices.h> #include <CoreServices/CoreServices.h>
#include <libkern/OSAtomic.h>
#include<boost/lockfree/spsc_queue.hpp> #include<boost/lockfree/spsc_queue.hpp>
struct OSXOutput { struct OSXOutput {
@ -48,12 +48,7 @@ struct OSXOutput {
AudioComponentInstance au; AudioComponentInstance au;
AudioStreamBasicDescription asbd; AudioStreamBasicDescription asbd;
Mutex mutex;
Cond condition;
boost::lockfree::spsc_queue<uint8_t> *ring_buffer; boost::lockfree::spsc_queue<uint8_t> *ring_buffer;
size_t render_buffer_size;
uint8_t *render_buffer;
OSXOutput() OSXOutput()
:base(osx_output_plugin) {} :base(osx_output_plugin) {}
@ -332,6 +327,56 @@ osx_output_sync_device_sample_rate(AudioDeviceID dev_id, AudioStreamBasicDescrip
} }
} }
static OSStatus
osx_output_set_buffer_size(AudioUnit au, AudioStreamBasicDescription desc, UInt32 *frame_size)
{
AudioValueRange value_range = {0, 0};
UInt32 property_size = sizeof(AudioValueRange);
OSStatus err = AudioUnitGetProperty(au,
kAudioDevicePropertyBufferFrameSizeRange,
kAudioUnitScope_Global,
0,
&value_range,
&property_size);
if (err != noErr)
return err;
UInt32 buffer_frame_size = value_range.mMaximum;
err = AudioUnitSetProperty(au,
kAudioDevicePropertyBufferFrameSize,
kAudioUnitScope_Global,
0,
&buffer_frame_size,
sizeof(buffer_frame_size));
if (err != noErr)
FormatWarning(osx_output_domain,
"Failed to set maximum buffer size: %d",
err);
property_size = sizeof(buffer_frame_size);
err = AudioUnitGetProperty(au,
kAudioDevicePropertyBufferFrameSize,
kAudioUnitScope_Global,
0,
&buffer_frame_size,
&property_size);
if (err != noErr) {
FormatWarning(osx_output_domain,
"Cannot get the buffer frame size: %d",
err);
return err;
}
buffer_frame_size *= desc.mBytesPerFrame;
// We set the frame size to a power of two integer that
// is larger than buffer_frame_size.
while (*frame_size < buffer_frame_size + 1) {
*frame_size <<= 1;
}
return noErr;
}
static void static void
osx_output_hog_device(AudioDeviceID dev_id, bool hog) osx_output_hog_device(AudioDeviceID dev_id, bool hog)
@ -504,18 +549,6 @@ done:
OS X audio subsystem (CoreAudio) to request audio data that will be OS X audio subsystem (CoreAudio) to request audio data that will be
played by the audio hardware. This function has hard time constraints played by the audio hardware. This function has hard time constraints
so it cannot do IO (debug statements) or memory allocations. so it cannot do IO (debug statements) or memory allocations.
The caller (i.e. CoreAudio) requests a specific number of
audio frames (in_number_frames) to be rendered into a
collection of output buffers (buffer_list). Depending on the
number of output buffers the render callback has to interleave
or de-interleave audio data to match the layout of the output
buffers. The intput buffer is always interleaved. In practice,
it seems this callback always gets a single output buffer
meaning that no de-interleaving actually takes place. For the
sake of correctness this callback allows for de-interleaving
anyway, and calculates the expected output layout by examining
the output buffers.
*/ */
static OSStatus static OSStatus
@ -526,79 +559,13 @@ osx_render(void *vdata,
UInt32 in_number_frames, UInt32 in_number_frames,
AudioBufferList *buffer_list) AudioBufferList *buffer_list)
{ {
AudioBuffer *output_buffer = nullptr;
size_t output_buffer_frame_size;
OSXOutput *od = (OSXOutput *) vdata; OSXOutput *od = (OSXOutput *) vdata;
/* int count = in_number_frames * od->asbd.mBytesPerFrame;
By convention when interfacing with audio hardware in CoreAudio, buffer_list->mBuffers[0].mDataByteSize =
in_bus_number equals 0 for output and 1 for input. Because this is an od->ring_buffer->pop((uint8_t *)buffer_list->mBuffers[0].mData,
audio output plugin our in_bus_number should always be 0. count);
*/ return noErr;
assert(in_bus_number == 0);
unsigned int input_channel_count = od->asbd.mChannelsPerFrame;
unsigned int output_channel_count = 0;
for (unsigned int i = 0 ; i < buffer_list->mNumberBuffers; ++i) {
output_buffer = &buffer_list->mBuffers[i];
assert(output_buffer->mData != nullptr);
output_channel_count += output_buffer->mNumberChannels;
}
assert(output_channel_count == input_channel_count);
size_t input_buffer_frame_size = od->asbd.mBytesPerFrame;
size_t sample_size = input_buffer_frame_size / input_channel_count;
size_t requested_bytes = in_number_frames * input_buffer_frame_size;
if (requested_bytes > od->render_buffer_size)
requested_bytes = od->render_buffer_size;
size_t available_bytes = od->ring_buffer->pop(od->render_buffer, requested_bytes);
/*
Maybe this is paranoid but we have no way of knowing
if the 'pop' above ended at a frame boundary. In case
of an incomplete last frame, keep popping until the
last frame is complete.
*/
while (true) {
size_t incomplete_frame_bytes = available_bytes % input_buffer_frame_size;
if (incomplete_frame_bytes == 0)
break;
available_bytes += od->ring_buffer->pop(
od->render_buffer + available_bytes,
input_buffer_frame_size - incomplete_frame_bytes
);
}
od->condition.signal(); // We are done consuming from ring_buffer
UInt32 available_frames = available_bytes / input_buffer_frame_size;
/*
To de-interleave the data in the input buffer so that it fits in
the output buffers we divide the input buffer frames into 'sub frames'
that fit into the output buffers.
*/
size_t sub_frame_offset = 0;
for (unsigned int i = 0 ; i < buffer_list->mNumberBuffers; ++i) {
output_buffer = &buffer_list->mBuffers[i];
output_buffer_frame_size = output_buffer->mNumberChannels * sample_size;
output_buffer->mDataByteSize = 0; // Record how much data we actually rendered
for (UInt32 current_frame = 0; current_frame < available_frames; ++current_frame) {
memcpy(
(uint8_t *) output_buffer->mData + current_frame * output_buffer_frame_size,
od->render_buffer + current_frame * input_buffer_frame_size + sub_frame_offset,
output_buffer_frame_size
);
output_buffer->mDataByteSize += output_buffer_frame_size;
}
sub_frame_offset += output_buffer_frame_size;
}
return noErr;
} }
static bool static bool
@ -635,24 +602,8 @@ osx_output_enable(AudioOutput *ao, Error &error)
return false; return false;
} }
if (oo->hog_device) { if (oo->hog_device) {
osx_output_hog_device(oo->dev_id, true); osx_output_hog_device(oo->dev_id, true);
}
AURenderCallbackStruct callback;
callback.inputProc = osx_render;
callback.inputProcRefCon = oo;
status =
AudioUnitSetProperty(oo->au,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0,
&callback, sizeof(callback));
if (status != noErr) {
AudioComponentInstanceDispose(oo->au);
error.Set(osx_output_domain, status,
"unable to set callback for OS X audio unit");
return false;
} }
return true; return true;
@ -665,9 +616,9 @@ osx_output_disable(AudioOutput *ao)
AudioComponentInstanceDispose(oo->au); AudioComponentInstanceDispose(oo->au);
if (oo->hog_device) { if (oo->hog_device) {
osx_output_hog_device(oo->dev_id, false); osx_output_hog_device(oo->dev_id, false);
} }
} }
static void static void
@ -679,7 +630,6 @@ osx_output_close(AudioOutput *ao)
AudioUnitUninitialize(od->au); AudioUnitUninitialize(od->au);
delete od->ring_buffer; delete od->ring_buffer;
delete[] od->render_buffer;
} }
static bool static bool
@ -736,6 +686,22 @@ osx_output_open(AudioOutput *ao, AudioFormat &audio_format,
return false; return false;
} }
AURenderCallbackStruct callback;
callback.inputProc = osx_render;
callback.inputProcRefCon = od;
status =
AudioUnitSetProperty(od->au,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0,
&callback, sizeof(callback));
if (status != noErr) {
AudioComponentInstanceDispose(od->au);
error.Set(osx_output_domain, status,
"unable to set callback for OS X audio unit");
return false;
}
status = AudioUnitInitialize(od->au); status = AudioUnitInitialize(od->au);
if (status != noErr) { if (status != noErr) {
osx_os_status_to_cstring(status, errormsg, sizeof(errormsg)); osx_os_status_to_cstring(status, errormsg, sizeof(errormsg));
@ -745,16 +711,17 @@ osx_output_open(AudioOutput *ao, AudioFormat &audio_format,
return false; return false;
} }
/* create a ring buffer of 1s */ UInt32 buffer_frame_size;
od->ring_buffer = new boost::lockfree::spsc_queue<uint8_t>(audio_format.sample_rate * audio_format.GetFrameSize()); status = osx_output_set_buffer_size(od->au, od->asbd, &buffer_frame_size);
if (status != noErr) {
osx_os_status_to_cstring(status, errormsg, sizeof(errormsg));
error.Format(osx_output_domain, status,
"Unable to set frame size: %s",
errormsg);
return false;
}
/* od->ring_buffer = new boost::lockfree::spsc_queue<uint8_t>(buffer_frame_size);
od->render_buffer_size is the maximum amount of data we
render in the render callback. Allocate enough space
for 0.1 s of frames.
*/
od->render_buffer_size = (audio_format.sample_rate/10) * audio_format.GetFrameSize();
od->render_buffer = new uint8_t[od->render_buffer_size];
status = AudioOutputUnitStart(od->au); status = AudioOutputUnitStart(od->au);
if (status != 0) { if (status != 0) {
@ -774,20 +741,13 @@ osx_output_play(AudioOutput *ao, const void *chunk, size_t size,
gcc_unused Error &error) gcc_unused Error &error)
{ {
OSXOutput *od = (OSXOutput *)ao; OSXOutput *od = (OSXOutput *)ao;
while (!od->ring_buffer->write_available()) {
{ struct timespec req;
const ScopeLock protect(od->mutex); req.tv_sec = 0;
req.tv_nsec = 25 * 1e6;
while (true) { nanosleep(&req, NULL);
if (od->ring_buffer->write_available() > 0)
break;
/* wait for some free space in the buffer */
od->condition.wait(od->mutex);
}
} }
return od->ring_buffer->push((uint8_t *)chunk, size);
return od->ring_buffer->push((uint8_t *) chunk, size);
} }
const struct AudioOutputPlugin osx_output_plugin = { const struct AudioOutputPlugin osx_output_plugin = {