diff options
| -rw-r--r-- | audio/out/ao.c | 4 | ||||
| -rw-r--r-- | audio/out/ao_coreaudio.c | 510 | ||||
| -rw-r--r-- | audio/out/ao_coreaudio_device.c | 449 | ||||
| -rw-r--r-- | audio/out/ao_coreaudio_utils.c | 62 | ||||
| -rw-r--r-- | audio/out/ao_coreaudio_utils.h | 3 | ||||
| -rw-r--r-- | wscript_build.py | 1 |
6 files changed, 596 insertions, 433 deletions
diff --git a/audio/out/ao.c b/audio/out/ao.c index 378197bcb3..977b8eb69a 100644 --- a/audio/out/ao.c +++ b/audio/out/ao.c @@ -38,6 +38,7 @@ extern const struct ao_driver audio_out_oss; extern const struct ao_driver audio_out_coreaudio; +extern const struct ao_driver audio_out_coreaudio_exclusive; extern const struct ao_driver audio_out_rsound; extern const struct ao_driver audio_out_sndio; extern const struct ao_driver audio_out_pulse; @@ -94,6 +95,9 @@ static const struct ao_driver * const audio_out_drivers[] = { #if HAVE_ENCODING &audio_out_lavc, #endif +#if HAVE_COREAUDIO + &audio_out_coreaudio_exclusive, +#endif #if HAVE_RSOUND &audio_out_rsound, #endif diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c index 1d44c81124..63009f3314 100644 --- a/audio/out/ao_coreaudio.c +++ b/audio/out/ao_coreaudio.c @@ -1,38 +1,18 @@ /* - * CoreAudio audio output driver for Mac OS X + * This file is part of mpv. * - * original copyright (C) Timothy J. Wood - Aug 2000 - * ported to MPlayer libao2 by Dan Christiansen - * - * Chris Roccati - * Stefano Pigozzi - * - * The S/PDIF part of the code is based on the auhal audio output - * module from VideoLAN: - * Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org> - * - * This file is part of MPlayer. - * - * MPlayer is free software; you can redistribute it and/or modify + * mpv is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * - * MPlayer is distributed in the hope that it will be useful, + * mpv is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along - * along with MPlayer; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/* - * The MacOS X CoreAudio framework doesn't mesh as simply as some - * simpler frameworks do. This is due to the fact that CoreAudio pulls - * audio samples rather than having them pushed at it (which is nice - * when you are wanting to do good buffering of audio). + * with mpv. If not, see <http://www.gnu.org/licenses/>. */ #include "config.h" @@ -50,47 +30,11 @@ static void audio_pause(struct ao *ao); static void audio_resume(struct ao *ao); static void reset(struct ao *ao); -static void print_buffer(struct ao *ao, struct mp_ring *buffer) -{ - void *tctx = talloc_new(NULL); - MP_VERBOSE(ao, "%s\n", mp_ring_repr(buffer, tctx)); - talloc_free(tctx); -} - -struct priv_d { - // 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; - AudioStreamBasicDescription original_asbd; - - bool changed_mixing; - int stream_asbd_changed; - bool muted; -}; - struct priv { AudioDeviceID device; // selected device - bool is_digital; // running in digital mode? - AudioUnit audio_unit; // AudioUnit for lpcm output - bool paused; - struct mp_ring *buffer; - struct priv_d *digital; // options int opt_device_id; @@ -124,77 +68,29 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags, return noErr; } -static OSStatus render_cb_digital( - AudioDeviceID device, const AudioTimeStamp *ts, - const void *in_data, const AudioTimeStamp *in_ts, - AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx) -{ - struct ao *ao = ctx; - struct priv *p = ao->priv; - struct priv_d *d = p->digital; - AudioBuffer buf = out_data->mBuffers[d->stream_idx]; - int requested = buf.mDataByteSize; - - if (d->muted) - mp_ring_drain(p->buffer, requested); - else - mp_ring_read(p->buffer, buf.mData, requested); - - return noErr; -} - static int control(struct ao *ao, enum aocontrol cmd, void *arg) { struct priv *p = ao->priv; - ao_control_vol_t *control_vol; - OSStatus err; - Float32 vol; switch (cmd) { - case AOCONTROL_GET_VOLUME: - control_vol = (ao_control_vol_t *)arg; - if (p->is_digital) { - struct priv_d *d = p->digital; - // Digital output has no volume adjust. - int digitalvol = d->muted ? 0 : 100; - *control_vol = (ao_control_vol_t) { - .left = digitalvol, .right = digitalvol, - }; - return CONTROL_TRUE; - } - - err = AudioUnitGetParameter(p->audio_unit, kHALOutputParam_Volume, - kAudioUnitScope_Global, 0, &vol); - + case AOCONTROL_GET_VOLUME: { + ao_control_vol_t *control_vol = (ao_control_vol_t *)arg; + float vol; + OSStatus err = + AudioUnitGetParameter(p->audio_unit, kHALOutputParam_Volume, + kAudioUnitScope_Global, 0, &vol); CHECK_CA_ERROR("could not get HAL output volume"); control_vol->left = control_vol->right = vol * 100.0; return CONTROL_TRUE; - - case AOCONTROL_SET_VOLUME: - control_vol = (ao_control_vol_t *)arg; - - 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, - // that's not we expected and the filter not support ac3 stream - // will cause mplayer die. - - // 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->muted = true; - else - d->muted = false; - return CONTROL_TRUE; - } - - vol = (control_vol->left + control_vol->right) / 200.0; - err = AudioUnitSetParameter(p->audio_unit, kHALOutputParam_Volume, - kAudioUnitScope_Global, 0, vol, 0); - + } + case AOCONTROL_SET_VOLUME: { + ao_control_vol_t *control_vol = (ao_control_vol_t *)arg; + float vol = (control_vol->left + control_vol->right) / 200.0; + OSStatus err = + AudioUnitSetParameter(p->audio_unit, kHALOutputParam_Volume, + kAudioUnitScope_Global, 0, vol, 0); CHECK_CA_ERROR("could not set HAL output volume"); return CONTROL_TRUE; - + } } // end switch return CONTROL_UNKNOWN; @@ -202,136 +98,30 @@ coreaudio_error: return CONTROL_ERROR; } -static void print_list(struct ao *ao) -{ - char *help = talloc_strdup(NULL, "Available output devices:\n"); - - AudioDeviceID *devs; - size_t n_devs; - - OSStatus err = - CA_GET_ARY(kAudioObjectSystemObject, kAudioHardwarePropertyDevices, - &devs, &n_devs); - - CHECK_CA_ERROR("Failed to get list of output devices."); - - for (int i = 0; i < n_devs; i++) { - char *name; - err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &name); - - if (err == noErr) - talloc_steal(devs, name); - else - name = "Unknown"; - - help = talloc_asprintf_append( - help, " * %s (id: %" PRIu32 ")\n", name, devs[i]); - } - - talloc_free(devs); - -coreaudio_error: - MP_INFO(ao, "%s", help); - talloc_free(help); -} - +static bool init_chmap(struct ao *ao); static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd); -static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd); static int init(struct ao *ao) { - OSStatus err; - struct priv *p = ao->priv; - - if (p->opt_list) print_list(ao); - - struct priv_d *d = talloc_zero(p, struct priv_d); - - *d = (struct priv_d) { - .muted = false, - .stream_asbd_changed = 0, - .hog_pid = -1, - .stream = 0, - .stream_idx = -1, - .changed_mixing = false, - }; + struct priv *p = ao->priv; - p->digital = d; + if (p->opt_list) ca_print_device_list(ao); ao->per_application_mixer = true; ao->no_persistent_volume = true; - AudioDeviceID selected_device = 0; - if (p->opt_device_id < 0) { - // device not set by user, get the default one - err = CA_GET(kAudioObjectSystemObject, - kAudioHardwarePropertyDefaultOutputDevice, - &selected_device); - CHECK_CA_ERROR("could not get default audio device"); - } else { - selected_device = p->opt_device_id; - } - - if (mp_msg_test(ao->log, MSGL_V)) { - char *name; - err = CA_GET_STR(selected_device, kAudioObjectPropertyName, &name); - CHECK_CA_ERROR("could not get selected audio device name"); + OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device); + CHECK_CA_ERROR("failed to select device"); - MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n", - name, selected_device); - - talloc_free(name); - } - - // Save selected device id - p->device = selected_device; + if (!init_chmap(ao)) + goto coreaudio_error; ao->format = af_fmt_from_planar(ao->format); - bool supports_digital = false; - /* Probe whether device support S/PDIF stream output if input is AC3 stream. */ - if (AF_FORMAT_IS_AC3(ao->format)) { - if (ca_device_supports_digital(ao, selected_device)) - supports_digital = true; - } - - if (!supports_digital) { - AudioChannelLayout *layouts; - size_t n_layouts; - err = CA_GET_ARY_O(selected_device, - kAudioDevicePropertyPreferredChannelLayout, - &layouts, &n_layouts); - CHECK_CA_ERROR("could not get audio device prefered layouts"); - - struct mp_chmap_sel chmap_sel = {0}; - for (int i = 0; i < n_layouts; i++) { - struct mp_chmap chmap = {0}; - if (ca_layout_to_mp_chmap(ao, &layouts[i], &chmap)) - mp_chmap_sel_add_map(&chmap_sel, &chmap); - } - - talloc_free(layouts); - - if (ao->channels.num < 3) { - struct mp_chmap chmap; - mp_chmap_from_channels(&chmap, ao->channels.num); - mp_chmap_sel_add_map(&chmap_sel, &chmap); - } - - if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) { - MP_ERR(ao, "could not select a suitable channel map among the " - "hardware supported ones. Make sure to configure your " - "output device correctly in 'Audio MIDI Setup.app'\n"); - goto coreaudio_error; - } - - } // closes if (!supports_digital) - // Build ASBD for the input format AudioStreamBasicDescription asbd; asbd.mSampleRate = ao->samplerate; - asbd.mFormatID = supports_digital ? - kAudioFormat60958AC3 : kAudioFormatLinearPCM; + asbd.mFormatID = kAudioFormatLinearPCM; asbd.mChannelsPerFrame = ao->channels.num; asbd.mBitsPerChannel = af_fmt2bits(ao->format); asbd.mFormatFlags = kAudioFormatFlagIsPacked; @@ -352,15 +142,52 @@ static int init(struct ao *ao) ca_print_asbd(ao, "source format:", &asbd); - if (supports_digital) - return init_digital(ao, asbd); - else - return init_lpcm(ao, asbd); + return init_lpcm(ao, asbd); coreaudio_error: return CONTROL_ERROR; } +static bool init_chmap(struct ao *ao) +{ + struct priv *p = ao->priv; + OSStatus err; + AudioChannelLayout *layouts; + size_t n_layouts; + + err = CA_GET_ARY_O(p->device, + kAudioDevicePropertyPreferredChannelLayout, + &layouts, &n_layouts); + CHECK_CA_ERROR("could not get audio device prefered layouts"); + + struct mp_chmap_sel chmap_sel = {0}; + for (int i = 0; i < n_layouts; i++) { + struct mp_chmap chmap = {0}; + if (ca_layout_to_mp_chmap(ao, &layouts[i], &chmap)) + mp_chmap_sel_add_map(&chmap_sel, &chmap); + } + + talloc_free(layouts); + + if (ao->channels.num < 3) { + struct mp_chmap chmap; + mp_chmap_from_channels(&chmap, ao->channels.num); + mp_chmap_sel_add_map(&chmap_sel, &chmap); + } + + if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) { + MP_ERR(ao, "could not select a suitable channel map among the " + "hardware supported ones. Make sure to configure your " + "output device correctly in 'Audio MIDI Setup.app'\n"); + goto coreaudio_error; + } + + return true; + +coreaudio_error: + return false; +} + static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd) { OSStatus err; @@ -408,8 +235,6 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd) "can't link audio unit to selected device"); p->buffer = mp_ring_new(p, get_ring_size(ao)); - print_buffer(ao, p->buffer); - AURenderCallbackStruct render_cb = (AURenderCallbackStruct) { .inputProc = render_cb_lpcm, .inputProcRefCon = ao, @@ -434,153 +259,11 @@ coreaudio_error: return CONTROL_ERROR; } -static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd) -{ - struct priv *p = ao->priv; - struct priv_d *d = p->digital; - OSStatus err = noErr; - - uint32_t is_alive = 1; - err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive); - CHECK_CA_WARN("could not check whether device is alive"); - - if (!is_alive) - MP_WARN(ao , "device is not alive\n"); - - p->is_digital = 1; - - err = ca_lock_device(p->device, &d->hog_pid); - CHECK_CA_WARN("failed to set hogmode"); - - err = ca_disable_mixing(ao, p->device, &d->changed_mixing); - CHECK_CA_WARN("failed to disable mixing"); - - AudioStreamID *streams; - size_t n_streams; - - /* Get a list of all the streams on this device. */ - err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams, - &streams, &n_streams); - - CHECK_CA_ERROR("could not get number of streams"); - - for (int i = 0; i < n_streams && d->stream_idx < 0; i++) { - bool digital = ca_stream_supports_digital(ao, streams[i]); - - if (digital) { - err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat, - &d->original_asbd); - if (!CHECK_CA_WARN("could not get stream's physical format to " - "revert to, getting the next one")) - continue; - - AudioStreamRangedDescription *formats; - size_t n_formats; - - err = CA_GET_ARY(streams[i], - kAudioStreamPropertyAvailablePhysicalFormats, - &formats, &n_formats); - - if (!CHECK_CA_WARN("could not get number of stream formats")) - continue; // try next one - - int req_rate_format = -1; - int max_rate_format = -1; - - d->stream = streams[i]; - d->stream_idx = i; - - for (int j = 0; j < n_formats; j++) - if (ca_format_is_digital(formats[j].mFormat)) { - // select the digital format that has exactly the same - // samplerate. If an exact match cannot be found, select - // the format with highest samplerate as backup. - if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) { - req_rate_format = j; - break; - } else if (max_rate_format < 0 || - formats[j].mFormat.mSampleRate > - formats[max_rate_format].mFormat.mSampleRate) - max_rate_format = j; - } - - if (req_rate_format >= 0) - d->stream_asbd = formats[req_rate_format].mFormat; - else - d->stream_asbd = formats[max_rate_format].mFormat; - - talloc_free(formats); - } - } - - talloc_free(streams); - - if (d->stream_idx < 0) { - MP_WARN(ao , "can't find any digital output stream format\n"); - goto coreaudio_error; - } - - if (!ca_change_format(ao, d->stream, d->stream_asbd)) - goto coreaudio_error; - - void *changed = (void *) &(d->stream_asbd_changed); - err = ca_enable_device_listener(p->device, changed); - CHECK_CA_ERROR("cannot install format change listener during init"); - -#if BYTE_ORDER == BIG_ENDIAN - if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian)) -#else - /* tell mplayer that we need a byteswap on AC3 streams, */ - if (d->stream_asbd.mFormatID & kAudioFormat60958AC3) - ao->format = AF_FORMAT_AC3_LE; - else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian) -#endif - MP_WARN(ao, "stream has non-native byte order, output may fail\n"); - - ao->samplerate = d->stream_asbd.mSampleRate; - ao->bps = ao->samplerate * - (d->stream_asbd.mBytesPerPacket / - d->stream_asbd.mFramesPerPacket); - - p->buffer = mp_ring_new(p, get_ring_size(ao)); - print_buffer(ao, p->buffer); - - err = AudioDeviceCreateIOProcID(p->device, - (AudioDeviceIOProc)render_cb_digital, - (void *)ao, - &d->render_cb); - - CHECK_CA_ERROR("failed to register digital render callback"); - - reset(ao); - - return CONTROL_TRUE; - -coreaudio_error: - err = ca_unlock_device(p->device, &d->hog_pid); - CHECK_CA_WARN("can't release hog mode"); - return CONTROL_ERROR; -} - static int play(struct ao *ao, void **data, int samples, int flags) { - struct priv *p = ao->priv; - struct priv_d *d = p->digital; + struct priv *p = ao->priv; void *output_samples = data[0]; - int num_bytes = samples * ao->sstride; - - // Check whether we need to reset the digital output stream. - if (p->is_digital && d->stream_asbd_changed) { - d->stream_asbd_changed = 0; - if (ca_stream_supports_digital(ao, d->stream)) { - if (!ca_change_format(ao, d->stream, d->stream_asbd)) { - MP_WARN(ao , "can't restore digital output\n"); - } else { - MP_WARN(ao, "restoring digital output succeeded.\n"); - reset(ao); - } - } - } + int num_bytes = samples * ao->sstride; int wrote = mp_ring_write(p->buffer, output_samples, num_bytes); audio_resume(ao); @@ -612,52 +295,20 @@ static float get_delay(struct ao *ao) static void uninit(struct ao *ao) { struct priv *p = ao->priv; - OSStatus err = noErr; - - if (!p->is_digital) { - AudioOutputUnitStop(p->audio_unit); - AudioUnitUninitialize(p->audio_unit); - AudioComponentInstanceDispose(p->audio_unit); - } else { - struct priv_d *d = p->digital; - - void *changed = (void *) &(d->stream_asbd_changed); - err = ca_disable_device_listener(p->device, changed); - CHECK_CA_WARN("can't remove device listener, this may cause a crash"); - - err = AudioDeviceStop(p->device, d->render_cb); - CHECK_CA_WARN("failed to stop audio device"); - - err = AudioDeviceDestroyIOProcID(p->device, d->render_cb); - CHECK_CA_WARN("failed to remove device render callback"); - - if (!ca_change_format(ao, d->stream, d->original_asbd)) - MP_WARN(ao, "can't revert to original device format"); - - err = ca_enable_mixing(ao, p->device, d->changed_mixing); - CHECK_CA_WARN("can't re-enable mixing"); - - err = ca_unlock_device(p->device, &d->hog_pid); - CHECK_CA_WARN("can't release hog mode"); - } + AudioOutputUnitStop(p->audio_unit); + AudioUnitUninitialize(p->audio_unit); + AudioComponentInstanceDispose(p->audio_unit); } static void audio_pause(struct ao *ao) { struct priv *p = ao->priv; - OSErr err = noErr; if (p->paused) return; - 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->device, d->render_cb); - CHECK_CA_WARN("can't stop digital device"); - } + OSStatus err = AudioOutputUnitStop(p->audio_unit); + CHECK_CA_WARN("can't stop audio unit"); p->paused = true; } @@ -665,19 +316,12 @@ static void audio_pause(struct ao *ao) static void audio_resume(struct ao *ao) { struct priv *p = ao->priv; - OSErr err = noErr; if (!p->paused) return; - 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->device, d->render_cb); - CHECK_CA_WARN("can't start digital device"); - } + OSStatus err = AudioOutputUnitStart(p->audio_unit); + CHECK_CA_WARN("can't start audio unit"); p->paused = false; } @@ -685,7 +329,7 @@ static void audio_resume(struct ao *ao) #define OPT_BASE_STRUCT struct priv const struct ao_driver audio_out_coreaudio = { - .description = "CoreAudio (OS X Audio Output)", + .description = "CoreAudio AudioUnit", .name = "coreaudio", .uninit = uninit, .init = init, diff --git a/audio/out/ao_coreaudio_device.c b/audio/out/ao_coreaudio_device.c new file mode 100644 index 0000000000..c86ffbd700 --- /dev/null +++ b/audio/out/ao_coreaudio_device.c @@ -0,0 +1,449 @@ +/* + * CoreAudio audio output driver for Mac OS X + * + * original copyright (C) Timothy J. Wood - Aug 2000 + * ported to MPlayer libao2 by Dan Christiansen + * + * Chris Roccati + * Stefano Pigozzi + * + * The S/PDIF part of the code is based on the auhal audio output + * module from VideoLAN: + * Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org> + * + * This file is part of MPlayer. + * + * MPlayer is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * MPlayer is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * along with MPlayer; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/* + * The MacOS X CoreAudio framework doesn't mesh as simply as some + * simpler frameworks do. This is due to the fact that CoreAudio pulls + * audio samples rather than having them pushed at it (which is nice + * when you are wanting to do good buffering of audio). + */ + +#include "config.h" +#include "ao.h" +#include "internal.h" +#include "audio/format.h" +#include "osdep/timer.h" +#include "options/m_option.h" +#include "misc/ring.h" +#include "common/msg.h" +#include "audio/out/ao_coreaudio_properties.h" +#include "audio/out/ao_coreaudio_utils.h" + +static void audio_pause(struct ao *ao); +static void audio_resume(struct ao *ao); +static void reset(struct ao *ao); + +struct priv { + AudioDeviceID device; // selected device + + bool paused; + + struct mp_ring *buffer; + + // 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; + AudioStreamBasicDescription original_asbd; + + bool changed_mixing; + int stream_asbd_changed; + bool muted; + + // options + int opt_device_id; + int opt_list; +}; + +static int get_ring_size(struct ao *ao) +{ + return af_fmt_seconds_to_bytes( + ao->format, 0.5, ao->channels.num, ao->samplerate); +} + +static OSStatus render_cb_digital( + AudioDeviceID device, const AudioTimeStamp *ts, + const void *in_data, const AudioTimeStamp *in_ts, + AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx) +{ + struct ao *ao = ctx; + struct priv *p = ao->priv; + AudioBuffer buf = out_data->mBuffers[p->stream_idx]; + int requested = buf.mDataByteSize; + + if (p->muted) + mp_ring_drain(p->buffer, requested); + else + mp_ring_read(p->buffer, buf.mData, requested); + + return noErr; +} + +static int control(struct ao *ao, enum aocontrol cmd, void *arg) +{ + struct priv *p = ao->priv; + ao_control_vol_t *control_vol; + switch (cmd) { + case AOCONTROL_GET_VOLUME: + control_vol = (ao_control_vol_t *)arg; + // Digital output has no volume adjust. + int digitalvol = p->muted ? 0 : 100; + *control_vol = (ao_control_vol_t) { + .left = digitalvol, .right = digitalvol, + }; + return CONTROL_TRUE; + + case AOCONTROL_SET_VOLUME: + control_vol = (ao_control_vol_t *)arg; + // Digital output can not set volume. Here we have to return true + // to make mixer forget it. Else mixer will add a soft filter, + // that's not we expected and the filter not support ac3 stream + // will cause mplayer die. + + // 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) + p->muted = true; + else + p->muted = false; + return CONTROL_TRUE; + + } // end switch + return CONTROL_UNKNOWN; +} + +static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd); + +static int init(struct ao *ao) +{ + struct priv *p = ao->priv; + + if (p->opt_list) ca_print_device_list(ao); + + *p = (struct priv) { + .muted = false, + .stream_asbd_changed = 0, + .hog_pid = -1, + .stream = 0, + .stream_idx = -1, + .changed_mixing = false, + }; + + OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device); + CHECK_CA_ERROR("failed to select device"); + + ao->format = af_fmt_from_planar(ao->format); + + bool supports_digital = false; + /* Probe whether device support S/PDIF stream output if input is AC3 stream. */ + if (AF_FORMAT_IS_AC3(ao->format)) { + if (ca_device_supports_digital(ao, p->device)) + supports_digital = true; + } + + if (!supports_digital) { + MP_ERR(ao, "selected device doesn't support digital formats\n"); + goto coreaudio_error; + } // closes if (!supports_digital) + + // Build ASBD for the input format + AudioStreamBasicDescription asbd; + asbd.mSampleRate = ao->samplerate; + asbd.mFormatID = kAudioFormat60958AC3; + asbd.mChannelsPerFrame = ao->channels.num; + asbd.mBitsPerChannel = af_fmt2bits(ao->format); + asbd.mFormatFlags = kAudioFormatFlagIsPacked; + + if ((ao->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F) + asbd.mFormatFlags |= kAudioFormatFlagIsFloat; + + if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI) + asbd.mFormatFlags |= kAudioFormatFlagIsSignedInteger; + + if ((ao->format & AF_FORMAT_END_MASK) == AF_FORMAT_BE) + asbd.mFormatFlags |= kAudioFormatFlagIsBigEndian; + + asbd.mFramesPerPacket = 1; + asbd.mBytesPerPacket = asbd.mBytesPerFrame = + asbd.mFramesPerPacket * asbd.mChannelsPerFrame * + (asbd.mBitsPerChannel / 8); + + ca_print_asbd(ao, "source format:", &asbd); + + return init_digital(ao, asbd); + +coreaudio_error: + return CONTROL_ERROR; +} + +static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd) +{ + struct priv *p = ao->priv; + OSStatus err = noErr; + + uint32_t is_alive = 1; + err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive); + CHECK_CA_WARN("could not check whether device is alive"); + + if (!is_alive) + MP_WARN(ao , "device is not alive\n"); + + err = ca_lock_device(p->device, &p->hog_pid); + CHECK_CA_WARN("failed to set hogmode"); + + err = ca_disable_mixing(ao, p->device, &p->changed_mixing); + CHECK_CA_WARN("failed to disable mixing"); + + AudioStreamID *streams; + size_t n_streams; + + /* Get a list of all the streams on this device. */ + err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams, + &streams, &n_streams); + + CHECK_CA_ERROR("could not get number of streams"); + + for (int i = 0; i < n_streams && p->stream_idx < 0; i++) { + bool digital = ca_stream_supports_digital(ao, streams[i]); + + if (digital) { + err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat, + &p->original_asbd); + if (!CHECK_CA_WARN("could not get stream's physical format to " + "revert to, getting the next one")) + continue; + + AudioStreamRangedDescription *formats; + size_t n_formats; + + err = CA_GET_ARY(streams[i], + kAudioStreamPropertyAvailablePhysicalFormats, + &formats, &n_formats); + + if (!CHECK_CA_WARN("could not get number of stream formats")) + continue; // try next one + + int req_rate_format = -1; + int max_rate_format = -1; + + p->stream = streams[i]; + p->stream_idx = i; + + for (int j = 0; j < n_formats; j++) + if (ca_format_is_digital(formats[j].mFormat)) { + // select the digital format that has exactly the same + // samplerate. If an exact match cannot be found, select + // the format with highest samplerate as backup. + if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) { + req_rate_format = j; + break; + } else if (max_rate_format < 0 || + formats[j].mFormat.mSampleRate > + formats[max_rate_format].mFormat.mSampleRate) + max_rate_format = j; + } + + if (req_rate_format >= 0) + p->stream_asbd = formats[req_rate_format].mFormat; + else + p->stream_asbd = formats[max_rate_format].mFormat; + + talloc_free(formats); + } + } + + talloc_free(streams); + + if (p->stream_idx < 0) { + MP_WARN(ao , "can't find any digital output stream format\n"); + goto coreaudio_error; + } + + if (!ca_change_format(ao, p->stream, p->stream_asbd)) + goto coreaudio_error; + + void *changed = (void *) &(p->stream_asbd_changed); + err = ca_enable_device_listener(p->device, changed); + CHECK_CA_ERROR("cannot install format change listener during init"); + +#if BYTE_ORDER == BIG_ENDIAN + if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian)) +#else + /* tell mplayer that we need a byteswap on AC3 streams, */ + if (p->stream_asbd.mFormatID & kAudioFormat60958AC3) + ao->format = AF_FORMAT_AC3_LE; + else if (p->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian) +#endif + MP_WARN(ao, "stream has non-native byte order, output may fail\n"); + + ao->samplerate = p->stream_asbd.mSampleRate; + ao->bps = ao->samplerate * + (p->stream_asbd.mBytesPerPacket / + p->stream_asbd.mFramesPerPacket); + + p->buffer = mp_ring_new(p, get_ring_size(ao)); + + err = AudioDeviceCreateIOProcID(p->device, + (AudioDeviceIOProc)render_cb_digital, + (void *)ao, + &p->render_cb); + + CHECK_CA_ERROR("failed to register digital render callback"); + + reset(ao); + + return CONTROL_TRUE; + +coreaudio_error: + err = ca_unlock_device(p->device, &p->hog_pid); + CHECK_CA_WARN("can't release hog mode"); + return CONTROL_ERROR; +} + +static int play(struct ao *ao, void **data, int sam |