mirror/mpv
0
0
Fork 0
mirror of https://github.com/mpv-player/mpv.git synced 2024-09-20 12:02:23 +02:00
Code

ao_coreaudio: change private vars names to match mpv conventions

Browse Source
This commit is contained in:
Stefano Pigozzi 2013-06-27 08:29:48 +02:00
parent d9c0dc7733
commit d3fb585b58
1 changed files with 101 additions and 91 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

192
audio/out/ao_coreaudio.c
View File

@ -54,24 +54,34 @@ static void print_buffer(struct mp_ring *buffer)
}
struct priv_d {
AudioDeviceIOProcID renderCallback; /* Render callback used for SPDIF */
pid_t i_hog_pid; /* Keeps the pid of our hog status. */
AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */
int i_stream_index; /* The index of i_stream_id in an AudioBufferList */
AudioStreamBasicDescription stream_format; /* The format we changed the stream to */
bool changed_mixing; /* Whether we need to set the mixing mode back */
int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */
int b_muted; /* Are we muted in digital mode? */
// digital render callback
AudioDeviceIOProcID render_cb;
// pid set for hog mode, (-1) means that hog mode on the device was
// released. hog mode is exclusive access to a device
pid_t hog_pid;
// stream selected for digital playback by the detection in init
AudioStreamID stream;
// stream index in an AudioBufferList
int stream_idx;
// format we changed the stream to: for the digital case each application
// sets the stream format for a device to what it needs
AudioStreamBasicDescription stream_asbd;
bool changed_mixing;
int stream_asbd_changed;
bool muted;
};
struct priv
{
AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */
int b_supports_digital; /* Does the currently selected device support digital mode? */
int b_digital; /* Are we running in digital mode? */
struct priv {
AudioDeviceID device; // selected device
bool supports_digital; // selected device supports digital mode?
bool is_digital; // running in digital mode?
/* AudioUnit */
AudioUnit theOutputUnit;
AudioUnit audio_unit; // AudioUnit for lpcm output
bool paused;
@ -108,10 +118,10 @@ static OSStatus render_cb_digital(
struct ao *ao = ctx;
struct priv *p = ao->priv;
struct priv_d *d = p->digital;
AudioBuffer buf = out_data->mBuffers[d->i_stream_index];
AudioBuffer buf = out_data->mBuffers[d->stream_idx];
int requested = buf.mDataByteSize;
if (d->b_muted)
if (d->muted)
mp_ring_drain(p->buffer, requested);
else
mp_ring_read(p->buffer, buf.mData, requested);
@ -128,17 +138,17 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
switch (cmd) {
case AOCONTROL_GET_VOLUME:
control_vol = (ao_control_vol_t *)arg;
if (p->b_digital) {
if (p->is_digital) {
struct priv_d *d = p->digital;
// Digital output has no volume adjust.
int vol = d->b_muted ? 0 : 100;
int vol = d->muted ? 0 : 100;
*control_vol = (ao_control_vol_t) {
.left = vol, .right = vol,
};
return CONTROL_TRUE;
}
err = AudioUnitGetParameter(p->theOutputUnit, kHALOutputParam_Volume,
err = AudioUnitGetParameter(p->audio_unit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, &vol);
CHECK_CA_ERROR("could not get HAL output volume");
@ -148,7 +158,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
case AOCONTROL_SET_VOLUME:
control_vol = (ao_control_vol_t *)arg;
if (p->b_digital) {
if (p->is_digital) {
struct priv_d *d = p->digital;
// Digital output can not set volume. Here we have to return true
// to make mixer forget it. Else mixer will add a soft filter,
@ -158,14 +168,14 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
// Although not support set volume, but at least we support mute.
// MPlayer set mute by set volume to zero, we handle it.
if (control_vol->left == 0 && control_vol->right == 0)
d->b_muted = 1;
d->muted = true;
else
d->b_muted = 0;
d->muted = false;
return CONTROL_TRUE;
}
vol = (control_vol->left + control_vol->right) * 4.0 / 200.0;
err = AudioUnitSetParameter(p->theOutputUnit, kHALOutputParam_Volume,
err = AudioUnitSetParameter(p->audio_unit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, vol, 0);
CHECK_CA_ERROR("could not set HAL output volume");
@ -178,7 +188,7 @@ coreaudio_error:
return CONTROL_ERROR;
}
static int AudioStreamChangeFormat(AudioStreamID i_stream_id,
static int AudioStreamChangeFormat(AudioStreamID stream,
AudioStreamBasicDescription change_format);
static void print_help(void)
@ -246,18 +256,18 @@ static int init(struct ao *ao, char *params)
struct priv *p = talloc_zero(ao, struct priv);
*p = (struct priv) {
.i_selected_dev = 0,
.b_supports_digital = 0,
.b_digital = 0,
.device = 0,
.supports_digital = false,
.is_digital = 0,
};
struct priv_d *d= talloc_zero(p, struct priv_d);
*d = (struct priv_d) {
.b_muted = 0,
.b_stream_format_changed = 0,
.i_hog_pid = -1,
.i_stream_id = 0,
.i_stream_index = -1,
.muted = false,
.stream_asbd_changed = 0,
.hog_pid = -1,
.stream = 0,
.stream_idx = -1,
.changed_mixing = false,
};
@ -292,7 +302,7 @@ static int init(struct ao *ao, char *params)
free(device_name);
// Save selected device id
p->i_selected_dev = selected_device;
p->device = selected_device;
struct mp_chmap_sel chmap_sel = {0};
mp_chmap_sel_add_waveext(&chmap_sel);
@ -302,7 +312,7 @@ static int init(struct ao *ao, char *params)
// Build ASBD for the input format
AudioStreamBasicDescription asbd;
asbd.mSampleRate = ao->samplerate;
asbd.mFormatID = p->b_supports_digital ?
asbd.mFormatID = p->supports_digital ?
kAudioFormat60958AC3 : kAudioFormatLinearPCM;
asbd.mChannelsPerFrame = ao->channels.num;
asbd.mBitsPerChannel = af_fmt2bits(ao->format);
@ -327,10 +337,10 @@ static int init(struct ao *ao, char *params)
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if (AF_FORMAT_IS_AC3(ao->format)) {
if (AudioDeviceSupportsDigital(selected_device))
p->b_supports_digital = 1;
p->supports_digital = true;
}
if (p->b_supports_digital)
if (p->supports_digital)
return init_digital(ao, asbd);
else
return init_lpcm(ao, asbd);
@ -359,16 +369,16 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
goto coreaudio_error;
}
err = AudioComponentInstanceNew(comp, &(p->theOutputUnit));
err = AudioComponentInstanceNew(comp, &(p->audio_unit));
CHECK_CA_ERROR("unable to open audio component");
// Initialize AudioUnit
err = AudioUnitInitialize(p->theOutputUnit);
err = AudioUnitInitialize(p->audio_unit);
CHECK_CA_ERROR_L(coreaudio_error_component,
"unable to initialize audio unit");
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitSetProperty(p->theOutputUnit,
err = AudioUnitSetProperty(p->audio_unit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &asbd, size);
@ -376,10 +386,10 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
"unable to set the input format on the audio unit");
//Set the Current Device to the Default Output Unit.
err = AudioUnitSetProperty(p->theOutputUnit,
err = AudioUnitSetProperty(p->audio_unit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0, &p->i_selected_dev,
sizeof(p->i_selected_dev));
kAudioUnitScope_Global, 0, &p->device,
sizeof(p->device));
p->buffer = mp_ring_new(p, get_ring_size(ao));
@ -390,7 +400,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
.inputProcRefCon = ao,
};
err = AudioUnitSetProperty(p->theOutputUnit,
err = AudioUnitSetProperty(p->audio_unit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &render_cb,
sizeof(AURenderCallbackStruct));
@ -402,9 +412,9 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
return CONTROL_OK;
coreaudio_error_audiounit:
AudioUnitUninitialize(p->theOutputUnit);
AudioUnitUninitialize(p->audio_unit);
coreaudio_error_component:
AudioComponentInstanceDispose(p->theOutputUnit);
AudioComponentInstanceDispose(p->audio_unit);
coreaudio_error:
return CONTROL_FALSE;
}
@ -418,7 +428,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
uint32_t size;
uint32_t is_alive = 1;
err = GetAudioProperty(p->i_selected_dev,
err = GetAudioProperty(p->device,
kAudioDevicePropertyDeviceIsAlive,
sizeof(uint32_t), &is_alive);
@ -427,19 +437,19 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
if (!is_alive)
ca_msg(MSGL_WARN, "device is not alive\n");
d->stream_format = asbd;
d->stream_asbd = asbd;
p->b_digital = 1;
p->is_digital = 1;
err = ca_lock_device(p->i_selected_dev, &d->i_hog_pid);
err = ca_lock_device(p->device, &d->hog_pid);
CHECK_CA_WARN("failed to set hogmode");
err = ca_disable_mixing(p->i_selected_dev, &d->changed_mixing);
err = ca_disable_mixing(p->device, &d->changed_mixing);
CHECK_CA_WARN("failed to disable mixing");
AudioStreamID *streams = NULL;
/* Get a list of all the streams on this device. */
size = GetAudioPropertyArray(p->i_selected_dev,
size = GetAudioPropertyArray(p->device,
kAudioDevicePropertyStreams,
kAudioDevicePropertyScopeOutput,
(void **)&streams);
@ -452,7 +462,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
int streams_n = size / sizeof(AudioStreamID);
// TODO: ++i is quite fishy in here. Investigate!
for (int i = 0; i < streams_n && d->i_stream_index < 0; ++i) {
for (int i = 0; i < streams_n && d->stream_idx < 0; ++i) {
bool digital = AudioStreamSupportsDigital(streams[i]);
if (digital) {
@ -472,8 +482,8 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
int req_rate_format = -1;
int max_rate_format = -1;
d->i_stream_id = streams[i];
d->i_stream_index = i;
d->stream = streams[i];
d->stream_idx = i;
// TODO: ++j is fishy. was like this in the original code. Investigate!
for (int j = 0; j < formats_n; ++j)
@ -482,7 +492,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
// samplerate. If an exact match cannot be found, select
// the format with highest samplerate as backup.
if (formats[j].mFormat.mSampleRate ==
d->stream_format.mSampleRate) {
d->stream_asbd.mSampleRate) {
req_rate_format = j;
break;
} else if (max_rate_format < 0 ||
@ -492,9 +502,9 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
}
if (req_rate_format >= 0)
d->stream_format = formats[req_rate_format].mFormat;
d->stream_asbd = formats[req_rate_format].mFormat;
else
d->stream_format = formats[max_rate_format].mFormat;
d->stream_asbd = formats[max_rate_format].mFormat;
free(formats);
}
@ -502,12 +512,12 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
free(streams);
if (d->i_stream_index < 0) {
if (d->stream_idx < 0) {
ca_msg(MSGL_WARN, "can't find any digital output stream format\n");
goto coreaudio_error;
}
if (!AudioStreamChangeFormat(d->i_stream_id, d->stream_format))
if (!AudioStreamChangeFormat(d->stream, d->stream_asbd))
goto coreaudio_error;
p_addr = (AudioObjectPropertyAddress) {
@ -516,37 +526,37 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
.mElement = kAudioObjectPropertyElementMaster,
};
const int *stream_format_changed = &(d->b_stream_format_changed);
err = AudioObjectAddPropertyListener(p->i_selected_dev,
const int *stream_asdb_changed = &(d->stream_asbd_changed);
err = AudioObjectAddPropertyListener(p->device,
&p_addr,
ca_device_listener,
(void *)stream_format_changed);
(void *)stream_asdb_changed);
CHECK_CA_ERROR("cannot install format change listener during init");
#if BYTE_ORDER == BIG_ENDIAN
if (!(p->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
/* tell mplayer that we need a byteswap on AC3 streams, */
if (d->stream_format.mFormatID & kAudioFormat60958AC3)
if (d->stream_asbd.mFormatID & kAudioFormat60958AC3)
ao->format = AF_FORMAT_AC3_LE;
else if (d->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
ca_msg(MSGL_WARN,
"stream has non-native byte order, digital output may fail\n");
#endif
ao->samplerate = d->stream_format.mSampleRate;
mp_chmap_from_channels(&ao->channels, d->stream_format.mChannelsPerFrame);
ao->samplerate = d->stream_asbd.mSampleRate;
mp_chmap_from_channels(&ao->channels, d->stream_asbd.mChannelsPerFrame);
ao->bps = ao->samplerate *
(d->stream_format.mBytesPerPacket /
d->stream_format.mFramesPerPacket);
(d->stream_asbd.mBytesPerPacket /
d->stream_asbd.mFramesPerPacket);
p->buffer = mp_ring_new(p, get_ring_size(ao));
print_buffer(p->buffer);
err = AudioDeviceCreateIOProcID(p->i_selected_dev,
err = AudioDeviceCreateIOProcID(p->device,
(AudioDeviceIOProc)render_cb_digital,
(void *)ao,
&d->renderCallback);
&d->render_cb);
CHECK_CA_ERROR("failed to register digital render callback");
@ -555,7 +565,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
return CONTROL_TRUE;
coreaudio_error:
err = ca_unlock_device(p->i_selected_dev, &d->i_hog_pid);
err = ca_unlock_device(p->device, &d->hog_pid);
CHECK_CA_WARN("can't release hog mode");
return CONTROL_FALSE;
}
@ -566,10 +576,10 @@ static int play(struct ao *ao, void *output_samples, int num_bytes, int flags)
struct priv_d *d = p->digital;
// Check whether we need to reset the digital output stream.
if (p->b_digital && d->b_stream_format_changed) {
d->b_stream_format_changed = 0;
if (AudioStreamSupportsDigital(d->i_stream_id)) {
if (!AudioStreamChangeFormat(d->i_stream_id, d->stream_format)) {
if (p->is_digital && d->stream_asbd_changed) {
d->stream_asbd_changed = 0;
if (AudioStreamSupportsDigital(d->stream)) {
if (!AudioStreamChangeFormat(d->stream, d->stream_asbd)) {
ca_msg(MSGL_WARN, "can't restore digital output\n");
} else {
ca_msg(MSGL_WARN, "restoring digital output succeeded.\n");
@ -613,23 +623,23 @@ static void uninit(struct ao *ao, bool immed)
if (!immed)
mp_sleep_us(get_delay(ao) * 1000000);
if (!p->b_digital) {
AudioOutputUnitStop(p->theOutputUnit);
AudioUnitUninitialize(p->theOutputUnit);
AudioComponentInstanceDispose(p->theOutputUnit);
if (!p->is_digital) {
AudioOutputUnitStop(p->audio_unit);
AudioUnitUninitialize(p->audio_unit);
AudioComponentInstanceDispose(p->audio_unit);
} else {
struct priv_d *d = p->digital;
err = AudioDeviceStop(p->i_selected_dev, d->renderCallback);
err = AudioDeviceStop(p->device, d->render_cb);
CHECK_CA_WARN("failed to stop audio device");
err = AudioDeviceDestroyIOProcID(p->i_selected_dev, d->renderCallback);
err = AudioDeviceDestroyIOProcID(p->device, d->render_cb);
CHECK_CA_WARN("failed to remove device render callback");
err = ca_enable_mixing(p->i_selected_dev, d->changed_mixing);
err = ca_enable_mixing(p->device, d->changed_mixing);
CHECK_CA_WARN("can't re-enable mixing");
err = ca_unlock_device(p->i_selected_dev, &d->i_hog_pid);
err = ca_unlock_device(p->device, &d->hog_pid);
CHECK_CA_WARN("can't release hog mode");
}
}
@ -642,12 +652,12 @@ static void audio_pause(struct ao *ao)
if (p->paused)
return;
if (!p->b_digital) {
err = AudioOutputUnitStop(p->theOutputUnit);
if (!p->is_digital) {
err = AudioOutputUnitStop(p->audio_unit);
CHECK_CA_WARN("can't stop audio unit");
} else {
struct priv_d *d = p->digital;
err = AudioDeviceStop(p->i_selected_dev, d->renderCallback);
err = AudioDeviceStop(p->device, d->render_cb);
CHECK_CA_WARN("can't stop digital device");
}
@ -662,12 +672,12 @@ static void audio_resume(struct ao *ao)
if (!p->paused)
return;
if (!p->b_digital) {
err = AudioOutputUnitStart(p->theOutputUnit);
if (!p->is_digital) {
err = AudioOutputUnitStart(p->audio_unit);
CHECK_CA_WARN("can't start audio unit");
} else {
struct priv_d *d = p->digital;
err = AudioDeviceStart(p->i_selected_dev, d->renderCallback);
err = AudioDeviceStart(p->device, d->render_cb);
CHECK_CA_WARN("can't start digital device");
}