mirror of
https://github.com/mpv-player/mpv
synced 2024-11-18 21:16:10 +01:00
ao_coreaudio: change private vars names to match mpv conventions
This commit is contained in:
Stefano Pigozzi
parent
d9c0dc7733
commit
d3fb585b58
@ -54,24 +54,34 @@ static void print_buffer(struct mp_ring *buffer)
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}
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struct priv_d {
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AudioDeviceIOProcID renderCallback; /* Render callback used for SPDIF */
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pid_t i_hog_pid; /* Keeps the pid of our hog status. */
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AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */
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int i_stream_index; /* The index of i_stream_id in an AudioBufferList */
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AudioStreamBasicDescription stream_format; /* The format we changed the stream to */
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bool changed_mixing; /* Whether we need to set the mixing mode back */
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int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */
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int b_muted; /* Are we muted in digital mode? */
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// digital render callback
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AudioDeviceIOProcID render_cb;
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// pid set for hog mode, (-1) means that hog mode on the device was
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// released. hog mode is exclusive access to a device
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pid_t hog_pid;
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// stream selected for digital playback by the detection in init
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AudioStreamID stream;
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// stream index in an AudioBufferList
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int stream_idx;
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// format we changed the stream to: for the digital case each application
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// sets the stream format for a device to what it needs
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AudioStreamBasicDescription stream_asbd;
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bool changed_mixing;
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int stream_asbd_changed;
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bool muted;
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};
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struct priv
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{
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AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */
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int b_supports_digital; /* Does the currently selected device support digital mode? */
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int b_digital; /* Are we running in digital mode? */
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struct priv {
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AudioDeviceID device; // selected device
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bool supports_digital; // selected device supports digital mode?
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bool is_digital; // running in digital mode?
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/* AudioUnit */
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AudioUnit theOutputUnit;
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AudioUnit audio_unit; // AudioUnit for lpcm output
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bool paused;
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@ -108,10 +118,10 @@ static OSStatus render_cb_digital(
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struct ao *ao = ctx;
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struct priv *p = ao->priv;
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struct priv_d *d = p->digital;
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AudioBuffer buf = out_data->mBuffers[d->i_stream_index];
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AudioBuffer buf = out_data->mBuffers[d->stream_idx];
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int requested = buf.mDataByteSize;
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if (d->b_muted)
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if (d->muted)
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mp_ring_drain(p->buffer, requested);
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else
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mp_ring_read(p->buffer, buf.mData, requested);
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@ -128,17 +138,17 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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control_vol = (ao_control_vol_t *)arg;
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if (p->b_digital) {
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if (p->is_digital) {
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struct priv_d *d = p->digital;
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// Digital output has no volume adjust.
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int vol = d->b_muted ? 0 : 100;
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int vol = d->muted ? 0 : 100;
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*control_vol = (ao_control_vol_t) {
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.left = vol, .right = vol,
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};
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return CONTROL_TRUE;
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}
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err = AudioUnitGetParameter(p->theOutputUnit, kHALOutputParam_Volume,
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err = AudioUnitGetParameter(p->audio_unit, kHALOutputParam_Volume,
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kAudioUnitScope_Global, 0, &vol);
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CHECK_CA_ERROR("could not get HAL output volume");
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@ -148,7 +158,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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case AOCONTROL_SET_VOLUME:
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control_vol = (ao_control_vol_t *)arg;
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if (p->b_digital) {
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if (p->is_digital) {
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struct priv_d *d = p->digital;
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// Digital output can not set volume. Here we have to return true
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// to make mixer forget it. Else mixer will add a soft filter,
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@ -158,14 +168,14 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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// Although not support set volume, but at least we support mute.
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// MPlayer set mute by set volume to zero, we handle it.
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if (control_vol->left == 0 && control_vol->right == 0)
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d->b_muted = 1;
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d->muted = true;
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else
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d->b_muted = 0;
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d->muted = false;
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return CONTROL_TRUE;
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}
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vol = (control_vol->left + control_vol->right) * 4.0 / 200.0;
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err = AudioUnitSetParameter(p->theOutputUnit, kHALOutputParam_Volume,
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err = AudioUnitSetParameter(p->audio_unit, kHALOutputParam_Volume,
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kAudioUnitScope_Global, 0, vol, 0);
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CHECK_CA_ERROR("could not set HAL output volume");
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@ -178,7 +188,7 @@ coreaudio_error:
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return CONTROL_ERROR;
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}
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static int AudioStreamChangeFormat(AudioStreamID i_stream_id,
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static int AudioStreamChangeFormat(AudioStreamID stream,
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AudioStreamBasicDescription change_format);
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static void print_help(void)
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@ -246,18 +256,18 @@ static int init(struct ao *ao, char *params)
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struct priv *p = talloc_zero(ao, struct priv);
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*p = (struct priv) {
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.i_selected_dev = 0,
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.b_supports_digital = 0,
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.b_digital = 0,
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.device = 0,
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.supports_digital = false,
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.is_digital = 0,
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};
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struct priv_d *d= talloc_zero(p, struct priv_d);
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*d = (struct priv_d) {
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.b_muted = 0,
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.b_stream_format_changed = 0,
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.i_hog_pid = -1,
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.i_stream_id = 0,
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.i_stream_index = -1,
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.muted = false,
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.stream_asbd_changed = 0,
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.hog_pid = -1,
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.stream = 0,
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.stream_idx = -1,
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.changed_mixing = false,
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};
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@ -292,7 +302,7 @@ static int init(struct ao *ao, char *params)
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free(device_name);
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// Save selected device id
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p->i_selected_dev = selected_device;
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p->device = selected_device;
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struct mp_chmap_sel chmap_sel = {0};
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mp_chmap_sel_add_waveext(&chmap_sel);
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@ -302,7 +312,7 @@ static int init(struct ao *ao, char *params)
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// Build ASBD for the input format
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AudioStreamBasicDescription asbd;
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asbd.mSampleRate = ao->samplerate;
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asbd.mFormatID = p->b_supports_digital ?
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asbd.mFormatID = p->supports_digital ?
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kAudioFormat60958AC3 : kAudioFormatLinearPCM;
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asbd.mChannelsPerFrame = ao->channels.num;
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asbd.mBitsPerChannel = af_fmt2bits(ao->format);
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@ -327,10 +337,10 @@ static int init(struct ao *ao, char *params)
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/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
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if (AF_FORMAT_IS_AC3(ao->format)) {
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if (AudioDeviceSupportsDigital(selected_device))
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p->b_supports_digital = 1;
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p->supports_digital = true;
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}
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if (p->b_supports_digital)
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if (p->supports_digital)
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return init_digital(ao, asbd);
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else
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return init_lpcm(ao, asbd);
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@ -359,16 +369,16 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
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goto coreaudio_error;
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}
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err = AudioComponentInstanceNew(comp, &(p->theOutputUnit));
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err = AudioComponentInstanceNew(comp, &(p->audio_unit));
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CHECK_CA_ERROR("unable to open audio component");
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// Initialize AudioUnit
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err = AudioUnitInitialize(p->theOutputUnit);
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err = AudioUnitInitialize(p->audio_unit);
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CHECK_CA_ERROR_L(coreaudio_error_component,
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"unable to initialize audio unit");
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size = sizeof(AudioStreamBasicDescription);
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err = AudioUnitSetProperty(p->theOutputUnit,
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err = AudioUnitSetProperty(p->audio_unit,
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kAudioUnitProperty_StreamFormat,
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kAudioUnitScope_Input, 0, &asbd, size);
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@ -376,10 +386,10 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
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"unable to set the input format on the audio unit");
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//Set the Current Device to the Default Output Unit.
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err = AudioUnitSetProperty(p->theOutputUnit,
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err = AudioUnitSetProperty(p->audio_unit,
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kAudioOutputUnitProperty_CurrentDevice,
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kAudioUnitScope_Global, 0, &p->i_selected_dev,
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sizeof(p->i_selected_dev));
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kAudioUnitScope_Global, 0, &p->device,
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sizeof(p->device));
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p->buffer = mp_ring_new(p, get_ring_size(ao));
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@ -390,7 +400,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
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.inputProcRefCon = ao,
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};
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err = AudioUnitSetProperty(p->theOutputUnit,
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err = AudioUnitSetProperty(p->audio_unit,
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kAudioUnitProperty_SetRenderCallback,
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kAudioUnitScope_Input, 0, &render_cb,
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sizeof(AURenderCallbackStruct));
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@ -402,9 +412,9 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
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return CONTROL_OK;
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coreaudio_error_audiounit:
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AudioUnitUninitialize(p->theOutputUnit);
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AudioUnitUninitialize(p->audio_unit);
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coreaudio_error_component:
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AudioComponentInstanceDispose(p->theOutputUnit);
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AudioComponentInstanceDispose(p->audio_unit);
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coreaudio_error:
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return CONTROL_FALSE;
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}
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@ -418,7 +428,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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uint32_t size;
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uint32_t is_alive = 1;
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err = GetAudioProperty(p->i_selected_dev,
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err = GetAudioProperty(p->device,
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kAudioDevicePropertyDeviceIsAlive,
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sizeof(uint32_t), &is_alive);
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@ -427,19 +437,19 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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if (!is_alive)
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ca_msg(MSGL_WARN, "device is not alive\n");
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d->stream_format = asbd;
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d->stream_asbd = asbd;
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p->b_digital = 1;
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p->is_digital = 1;
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err = ca_lock_device(p->i_selected_dev, &d->i_hog_pid);
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err = ca_lock_device(p->device, &d->hog_pid);
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CHECK_CA_WARN("failed to set hogmode");
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err = ca_disable_mixing(p->i_selected_dev, &d->changed_mixing);
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err = ca_disable_mixing(p->device, &d->changed_mixing);
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CHECK_CA_WARN("failed to disable mixing");
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AudioStreamID *streams = NULL;
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/* Get a list of all the streams on this device. */
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size = GetAudioPropertyArray(p->i_selected_dev,
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size = GetAudioPropertyArray(p->device,
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kAudioDevicePropertyStreams,
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kAudioDevicePropertyScopeOutput,
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(void **)&streams);
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@ -452,7 +462,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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int streams_n = size / sizeof(AudioStreamID);
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// TODO: ++i is quite fishy in here. Investigate!
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for (int i = 0; i < streams_n && d->i_stream_index < 0; ++i) {
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for (int i = 0; i < streams_n && d->stream_idx < 0; ++i) {
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bool digital = AudioStreamSupportsDigital(streams[i]);
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if (digital) {
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@ -472,8 +482,8 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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int req_rate_format = -1;
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int max_rate_format = -1;
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d->i_stream_id = streams[i];
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d->i_stream_index = i;
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d->stream = streams[i];
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d->stream_idx = i;
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// TODO: ++j is fishy. was like this in the original code. Investigate!
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for (int j = 0; j < formats_n; ++j)
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@ -482,7 +492,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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// samplerate. If an exact match cannot be found, select
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// the format with highest samplerate as backup.
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if (formats[j].mFormat.mSampleRate ==
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d->stream_format.mSampleRate) {
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d->stream_asbd.mSampleRate) {
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req_rate_format = j;
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break;
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} else if (max_rate_format < 0 ||
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@ -492,9 +502,9 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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}
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if (req_rate_format >= 0)
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d->stream_format = formats[req_rate_format].mFormat;
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d->stream_asbd = formats[req_rate_format].mFormat;
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else
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d->stream_format = formats[max_rate_format].mFormat;
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d->stream_asbd = formats[max_rate_format].mFormat;
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free(formats);
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}
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@ -502,12 +512,12 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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free(streams);
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if (d->i_stream_index < 0) {
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if (d->stream_idx < 0) {
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ca_msg(MSGL_WARN, "can't find any digital output stream format\n");
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goto coreaudio_error;
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}
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if (!AudioStreamChangeFormat(d->i_stream_id, d->stream_format))
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if (!AudioStreamChangeFormat(d->stream, d->stream_asbd))
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goto coreaudio_error;
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p_addr = (AudioObjectPropertyAddress) {
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@ -516,37 +526,37 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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.mElement = kAudioObjectPropertyElementMaster,
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};
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const int *stream_format_changed = &(d->b_stream_format_changed);
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err = AudioObjectAddPropertyListener(p->i_selected_dev,
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const int *stream_asdb_changed = &(d->stream_asbd_changed);
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err = AudioObjectAddPropertyListener(p->device,
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&p_addr,
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ca_device_listener,
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(void *)stream_format_changed);
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(void *)stream_asdb_changed);
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CHECK_CA_ERROR("cannot install format change listener during init");
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#if BYTE_ORDER == BIG_ENDIAN
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if (!(p->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
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if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian))
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#else
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/* tell mplayer that we need a byteswap on AC3 streams, */
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if (d->stream_format.mFormatID & kAudioFormat60958AC3)
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if (d->stream_asbd.mFormatID & kAudioFormat60958AC3)
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ao->format = AF_FORMAT_AC3_LE;
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else if (d->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
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else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
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ca_msg(MSGL_WARN,
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"stream has non-native byte order, digital output may fail\n");
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#endif
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ao->samplerate = d->stream_format.mSampleRate;
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mp_chmap_from_channels(&ao->channels, d->stream_format.mChannelsPerFrame);
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ao->samplerate = d->stream_asbd.mSampleRate;
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mp_chmap_from_channels(&ao->channels, d->stream_asbd.mChannelsPerFrame);
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ao->bps = ao->samplerate *
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(d->stream_format.mBytesPerPacket /
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d->stream_format.mFramesPerPacket);
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(d->stream_asbd.mBytesPerPacket /
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d->stream_asbd.mFramesPerPacket);
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p->buffer = mp_ring_new(p, get_ring_size(ao));
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print_buffer(p->buffer);
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err = AudioDeviceCreateIOProcID(p->i_selected_dev,
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err = AudioDeviceCreateIOProcID(p->device,
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(AudioDeviceIOProc)render_cb_digital,
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(void *)ao,
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&d->renderCallback);
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&d->render_cb);
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CHECK_CA_ERROR("failed to register digital render callback");
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@ -555,7 +565,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
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return CONTROL_TRUE;
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coreaudio_error:
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err = ca_unlock_device(p->i_selected_dev, &d->i_hog_pid);
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err = ca_unlock_device(p->device, &d->hog_pid);
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CHECK_CA_WARN("can't release hog mode");
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return CONTROL_FALSE;
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}
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@ -566,10 +576,10 @@ static int play(struct ao *ao, void *output_samples, int num_bytes, int flags)
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struct priv_d *d = p->digital;
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// Check whether we need to reset the digital output stream.
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if (p->b_digital && d->b_stream_format_changed) {
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d->b_stream_format_changed = 0;
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if (AudioStreamSupportsDigital(d->i_stream_id)) {
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if (!AudioStreamChangeFormat(d->i_stream_id, d->stream_format)) {
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if (p->is_digital && d->stream_asbd_changed) {
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d->stream_asbd_changed = 0;
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if (AudioStreamSupportsDigital(d->stream)) {
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if (!AudioStreamChangeFormat(d->stream, d->stream_asbd)) {
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ca_msg(MSGL_WARN, "can't restore digital output\n");
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} else {
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ca_msg(MSGL_WARN, "restoring digital output succeeded.\n");
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@ -613,23 +623,23 @@ static void uninit(struct ao *ao, bool immed)
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if (!immed)
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mp_sleep_us(get_delay(ao) * 1000000);
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if (!p->b_digital) {
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AudioOutputUnitStop(p->theOutputUnit);
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AudioUnitUninitialize(p->theOutputUnit);
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AudioComponentInstanceDispose(p->theOutputUnit);
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if (!p->is_digital) {
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AudioOutputUnitStop(p->audio_unit);
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AudioUnitUninitialize(p->audio_unit);
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AudioComponentInstanceDispose(p->audio_unit);
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} else {
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struct priv_d *d = p->digital;
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err = AudioDeviceStop(p->i_selected_dev, d->renderCallback);
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err = AudioDeviceStop(p->device, d->render_cb);
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CHECK_CA_WARN("failed to stop audio device");
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err = AudioDeviceDestroyIOProcID(p->i_selected_dev, d->renderCallback);
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err = AudioDeviceDestroyIOProcID(p->device, d->render_cb);
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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");
|
||||
}
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user