diff options
Diffstat (limited to 'audio/out/ao_coreaudio_exclusive.c')
| -rw-r--r-- | audio/out/ao_coreaudio_exclusive.c | 667 |
1 files changed, 667 insertions, 0 deletions
diff --git a/audio/out/ao_coreaudio_exclusive.c b/audio/out/ao_coreaudio_exclusive.c new file mode 100644 index 0000000000..49560a2e80 --- /dev/null +++ b/audio/out/ao_coreaudio_exclusive.c @@ -0,0 +1,667 @@ +/* + * 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); + +static bool ca_format_is_digital(AudioStreamBasicDescription asbd) +{ + switch (asbd.mFormatID) + case 'IAC3': + case 'iac3': + case kAudioFormat60958AC3: + case kAudioFormatAC3: + return true; + return false; +} + +static bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream) +{ + AudioStreamRangedDescription *formats = NULL; + size_t n_formats; + + OSStatus err = + CA_GET_ARY(stream, kAudioStreamPropertyAvailablePhysicalFormats, + &formats, &n_formats); + + CHECK_CA_ERROR("Could not get number of stream formats."); + + for (int i = 0; i < n_formats; i++) { + AudioStreamBasicDescription asbd = formats[i].mFormat; + ca_print_asbd(ao, "supported format:", &(asbd)); + if (ca_format_is_digital(asbd)) { + talloc_free(formats); + return true; + } + } + + talloc_free(formats); +coreaudio_error: + return false; +} + +static bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device) +{ + AudioStreamID *streams = NULL; + size_t n_streams; + + /* Retrieve all the output streams. */ + OSStatus err = + CA_GET_ARY_O(device, kAudioDevicePropertyStreams, &streams, &n_streams); + + CHECK_CA_ERROR("could not get number of streams."); + + for (int i = 0; i < n_streams; i++) { + if (ca_stream_supports_digital(ao, streams[i])) { + talloc_free(streams); + return true; + } + } + + talloc_free(streams); + +coreaudio_error: + return false; +} + +static OSStatus ca_property_listener( + AudioObjectPropertySelector selector, + AudioObjectID object, uint32_t n_addresses, + const AudioObjectPropertyAddress addresses[], + void *data) +{ + void *talloc_ctx = talloc_new(NULL); + + for (int i = 0; i < n_addresses; i++) { + if (addresses[i].mSelector == selector) { + if (data) *(volatile int *)data = 1; + break; + } + } + talloc_free(talloc_ctx); + return noErr; +} + +static OSStatus ca_stream_listener( + AudioObjectID object, uint32_t n_addresses, + const AudioObjectPropertyAddress addresses[], + void *data) +{ + return ca_property_listener(kAudioStreamPropertyPhysicalFormat, + object, n_addresses, addresses, data); +} + +static OSStatus ca_device_listener( + AudioObjectID object, uint32_t n_addresses, + const AudioObjectPropertyAddress addresses[], + void *data) +{ + return ca_property_listener(kAudioDevicePropertyDeviceHasChanged, + object, n_addresses, addresses, data); +} + +static OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid) { + *pid = getpid(); + OSStatus err = CA_SET(device, kAudioDevicePropertyHogMode, pid); + if (err != noErr) + *pid = -1; + + return err; +} + +static OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid) { + if (*pid == getpid()) { + *pid = -1; + return CA_SET(device, kAudioDevicePropertyHogMode, &pid); + } + return noErr; +} + +static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device, + uint32_t val, bool *changed) { + *changed = false; + + AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) { + .mSelector = kAudioDevicePropertySupportsMixing, + .mScope = kAudioObjectPropertyScopeGlobal, + .mElement = kAudioObjectPropertyElementMaster, + }; + + if (AudioObjectHasProperty(device, &p_addr)) { + OSStatus err; + Boolean writeable = 0; + err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing, + &writeable); + + if (!CHECK_CA_WARN("can't tell if mixing property is settable")) { + return err; + } + + if (!writeable) + return noErr; + + err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val); + if (err != noErr) + return err; + + if (!CHECK_CA_WARN("can't set mix mode")) { + return err; + } + + *changed = true; + } + + return noErr; +} + +static OSStatus ca_disable_mixing(struct ao *ao, + AudioDeviceID device, bool *changed) { + return ca_change_mixing(ao, device, 0, changed); +} + +static OSStatus ca_enable_mixing(struct ao *ao, + AudioDeviceID device, bool changed) { + if (changed) { + bool dont_care = false; + return ca_change_mixing(ao, device, 1, &dont_care); + } + + return noErr; +} + +static OSStatus ca_change_device_listening(AudioDeviceID device, + void *flag, bool enabled) +{ + AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) { + .mSelector = kAudioDevicePropertyDeviceHasChanged, + .mScope = kAudioObjectPropertyScopeGlobal, + .mElement = kAudioObjectPropertyElementMaster, + }; + + if (enabled) { + return AudioObjectAddPropertyListener( + device, &p_addr, ca_device_listener, flag); + } else { + return AudioObjectRemovePropertyListener( + device, &p_addr, ca_device_listener, flag); + } +} + +static OSStatus ca_enable_device_listener(AudioDeviceID device, void *flag) { + return ca_change_device_listening(device, flag, true); +} + +static OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag) { + return ca_change_device_listening(device, flag, false); +} + +static bool ca_change_format(struct ao *ao, AudioStreamID stream, + AudioStreamBasicDescription change_format) +{ + OSStatus err = noErr; + AudioObjectPropertyAddress p_addr; + volatile int stream_format_changed = 0; + + ca_print_asbd(ao, "setting stream format:", &change_format); + + /* Install the callback. */ + p_addr = (AudioObjectPropertyAddress) { + .mSelector = kAudioStreamPropertyPhysicalFormat, + .mScope = kAudioObjectPropertyScopeGlobal, + .mElement = kAudioObjectPropertyElementMaster, + }; + + err = AudioObjectAddPropertyListener(stream, &p_addr, ca_stream_listener, + (void *)&stream_format_changed); + if (!CHECK_CA_WARN("can't add property listener during format change")) { + return false; + } + + /* Change the format. */ + err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &change_format); + if (!CHECK_CA_WARN("error changing physical format")) { + return false; + } + + /* The AudioStreamSetProperty is not only asynchronious, + * it is also not Atomic, in its behaviour. + * Therefore we check 5 times before we really give up. */ + bool format_set = false; + for (int i = 0; !format_set && i < 5; i++) { + for (int j = 0; !stream_format_changed && j < 50; j++) + mp_sleep_us(10000); + + if (stream_format_changed) { + stream_format_changed = 0; + } else { + MP_VERBOSE(ao, "reached timeout\n"); + } + + AudioStreamBasicDescription actual_format; + err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format); + + ca_print_asbd(ao, "actual format in use:", &actual_format); + if (actual_format.mSampleRate == change_format.mSampleRate && + actual_format.mFormatID == change_format.mFormatID && + actual_format.mFramesPerPacket == change_format.mFramesPerPacket) { + format_set = true; + } + } + + err = AudioObjectRemovePropertyListener(stream, &p_addr, ca_stream_listener, + (void *)&stream_format_changed); + + if (!CHECK_CA_WARN("can't remove property listener")) { + return false; + } + + return format_set; +} + + +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; +}; + +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; + + OSStatus err = ca_select_device(ao, ao->device, &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, + * or anything else IEC61937-framed. */ + if (AF_FORMAT_IS_IEC61937(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; + ca_fill_asbd(ao, &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 (p->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian) + 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 samples, int flags) +{ + 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->stream_asbd_changed) { + p->stream_asbd_changed = 0; + if (ca_stream_supports_digital(ao, p->stream)) { + if (!ca_change_format(ao, p->stream, p->stream_asbd)) { + MP_WARN(ao , "can't restore digital output\n"); + } else { + MP_WARN(ao, "restoring digital output succeeded.\n"); + reset(ao); + } + } + } + + int wrote = mp_ring_write(p->buffer, output_samples, num_bytes); + audio_resume(ao); + + return wrote / ao->sstride; +} + +static void reset(struct ao *ao) +{ + struct priv *p = ao->priv; + audio_pause(ao); + mp_ring_reset(p->buffer); +} + +static int get_space(struct ao *ao) +{ + struct priv *p = ao->priv; + return mp_ring_available(p->buffer) / ao->sstride; +} + +static float get_delay(struct ao *ao) +{ + // FIXME: should also report the delay of coreaudio itself (hardware + + // internal buffers) + struct priv *p = ao->priv; + return mp_ring_buffered(p->buffer) / (float)ao->bps; +} + +static void uninit(struct ao *ao) +{ + struct priv *p = ao->priv; + OSStatus err = noErr; + + void *changed = (void *) &(p->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, p->render_cb); + CHECK_CA_WARN("failed to stop audio device"); + + err = AudioDeviceDestroyIOProcID(p->device, p->render_cb); + CHECK_CA_WARN("failed to remove device render callback"); + + if (!ca_change_format(ao, p->stream, p->original_asbd)) + MP_WARN(ao, "can't revert to original device format"); + + err = ca_enable_mixing(ao, p->device, p->changed_mixing); + CHECK_CA_WARN("can't re-enable mixing"); + + err = ca_unlock_device(p->device, &p->hog_pid); + CHECK_CA_WARN("can't release hog mode"); +} + +static void audio_pause(struct ao *ao) +{ + struct priv *p = ao->priv; + + if (p->paused) + return; + + OSStatus err = AudioDeviceStop(p->device, p->render_cb); + CHECK_CA_WARN("can't stop digital device"); + + p->paused = true; +} + +static void audio_resume(struct ao *ao) +{ + struct priv *p = ao->priv; + + if (!p->paused) + return; + + OSStatus err = AudioDeviceStart(p->device, p->render_cb); + CHECK_CA_WARN("can't start digital device"); + + p->paused = false; +} + +#define OPT_BASE_STRUCT struct priv + +const struct ao_driver audio_out_coreaudio_exclusive = { + .description = "CoreAudio Exclusive Mode", + .name = "coreaudio_exclusive", + .uninit = uninit, + .init = init, + .play = play, + .control = control, + .get_space = get_space, + .get_delay = get_delay, + .reset = reset, + .pause = audio_pause, + .resume = audio_resume, + .list_devs = ca_get_device_list, + .priv_size = sizeof(struct priv), + .priv_defaults = &(const struct priv){ + .muted = false, + .stream_asbd_changed = 0, + .hog_pid = -1, + .stream = 0, + .stream_idx = -1, + .changed_mixing = false, + }, +}; |