mirror of
https://github.com/mpv-player/mpv
synced 2024-12-25 16:33:02 +00:00
591e21a2eb
The standard header is stdatomic.h, so the extra "s" freaks me out every time I look at it.
413 lines
13 KiB
C
413 lines
13 KiB
C
/*
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* CoreAudio audio output driver for Mac OS X
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*
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* original copyright (C) Timothy J. Wood - Aug 2000
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* ported to MPlayer libao2 by Dan Christiansen
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*
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* Chris Roccati
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* Stefano Pigozzi
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*
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* The S/PDIF part of the code is based on the auhal audio output
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* module from VideoLAN:
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* Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
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*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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/*
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* The MacOS X CoreAudio framework doesn't mesh as simply as some
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* simpler frameworks do. This is due to the fact that CoreAudio pulls
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* audio samples rather than having them pushed at it (which is nice
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* when you are wanting to do good buffering of audio).
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*/
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#include <CoreAudio/HostTime.h>
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#include "config.h"
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#include "ao.h"
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#include "internal.h"
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#include "audio/format.h"
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#include "osdep/timer.h"
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#include "osdep/atomic.h"
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#include "options/m_option.h"
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#include "common/msg.h"
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#include "audio/out/ao_coreaudio_chmap.h"
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#include "audio/out/ao_coreaudio_properties.h"
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#include "audio/out/ao_coreaudio_utils.h"
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struct priv {
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AudioDeviceID device; // selected device
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bool paused;
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// audio 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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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, and the original format to restore
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AudioStreamBasicDescription stream_asbd;
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AudioStreamBasicDescription original_asbd;
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bool changed_mixing;
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atomic_bool reload_requested;
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uint32_t hw_latency_us;
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};
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static OSStatus property_listener_cb(
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AudioObjectID object, uint32_t n_addresses,
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const AudioObjectPropertyAddress addresses[],
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void *data)
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{
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struct ao *ao = data;
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struct priv *p = ao->priv;
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// Check whether we need to reset the compressed output stream.
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AudioStreamBasicDescription f;
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OSErr err = CA_GET(p->stream, kAudioStreamPropertyVirtualFormat, &f);
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CHECK_CA_WARN("could not get stream format");
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if (err != noErr || !ca_asbd_equals(&p->stream_asbd, &f)) {
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if (atomic_compare_exchange_strong(&p->reload_requested,
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&(bool){false}, true))
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{
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ao_request_reload(ao);
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MP_INFO(ao, "Stream format changed! Reloading.\n");
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}
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}
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return noErr;
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}
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static OSStatus enable_property_listener(struct ao *ao, bool enabled)
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{
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struct priv *p = ao->priv;
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uint32_t selectors[] = {kAudioDevicePropertyDeviceHasChanged,
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kAudioHardwarePropertyDevices};
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AudioDeviceID devs[] = {p->device,
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kAudioObjectSystemObject};
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assert(MP_ARRAY_SIZE(selectors) == MP_ARRAY_SIZE(devs));
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OSStatus status = noErr;
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for (int n = 0; n < MP_ARRAY_SIZE(devs); n++) {
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AudioObjectPropertyAddress addr = {
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.mScope = kAudioObjectPropertyScopeGlobal,
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.mElement = kAudioObjectPropertyElementMaster,
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.mSelector = selectors[n],
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};
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AudioDeviceID device = devs[n];
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OSStatus status2;
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if (enabled) {
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status2 = AudioObjectAddPropertyListener(
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device, &addr, property_listener_cb, ao);
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} else {
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status2 = AudioObjectRemovePropertyListener(
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device, &addr, property_listener_cb, ao);
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}
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if (status == noErr)
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status = status2;
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}
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return status;
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}
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static OSStatus render_cb_compressed(
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AudioDeviceID device, const AudioTimeStamp *ts,
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const void *in_data, const AudioTimeStamp *in_ts,
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AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx)
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{
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struct ao *ao = ctx;
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struct priv *p = ao->priv;
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AudioBuffer buf = out_data->mBuffers[p->stream_idx];
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int requested = buf.mDataByteSize;
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int pseudo_frames = requested / ao->sstride;
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// we expect the callback to read full frames, which are aligned accordingly
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if (pseudo_frames * ao->sstride != requested) {
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MP_ERR(ao, "Unsupported unaligned read of %d bytes.\n", requested);
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return kAudioHardwareUnspecifiedError;
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}
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int64_t end = mp_time_us();
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end += p->hw_latency_us + ca_get_latency(ts)
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+ ca_frames_to_us(ao, pseudo_frames);
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ao_read_data(ao, &buf.mData, pseudo_frames, end);
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return noErr;
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}
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// Apparently, audio devices can have multiple sub-streams. It's not clear to
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// me what devices with multiple streams actually do. So only select the first
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// one that fulfills some minimum requirements.
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// If this is not sufficient, we could duplicate the device list entries for
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// each sub-stream, and make it explicit.
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static int select_stream(struct ao *ao)
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{
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struct priv *p = ao->priv;
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AudioStreamID *streams;
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size_t n_streams;
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OSStatus err;
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/* Get a list of all the streams on this device. */
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err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
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&streams, &n_streams);
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CHECK_CA_ERROR("could not get number of streams");
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for (int i = 0; i < n_streams; i++) {
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uint32_t direction;
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err = CA_GET(streams[i], kAudioStreamPropertyDirection, &direction);
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CHECK_CA_WARN("could not get stream direction");
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if (err == noErr && direction != 0) {
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MP_VERBOSE(ao, "Substream %d is not an output stream.\n", i);
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continue;
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}
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if (af_fmt_is_pcm(ao->format) || ca_stream_supports_compressed(ao,
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streams[i]))
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{
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MP_VERBOSE(ao, "Using substream %d/%zd.\n", i, n_streams);
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p->stream = streams[i];
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p->stream_idx = i;
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break;
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}
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}
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talloc_free(streams);
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if (p->stream_idx < 0) {
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MP_ERR(ao, "No useable substream found.\n");
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goto coreaudio_error;
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}
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return 0;
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coreaudio_error:
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return -1;
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}
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static int find_best_format(struct ao *ao, AudioStreamBasicDescription *out_fmt)
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{
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struct priv *p = ao->priv;
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// Build ASBD for the input format
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AudioStreamBasicDescription asbd;
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ca_fill_asbd(ao, &asbd);
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ca_print_asbd(ao, "our format:", &asbd);
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*out_fmt = (AudioStreamBasicDescription){0};
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AudioStreamRangedDescription *formats;
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size_t n_formats;
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OSStatus err;
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err = CA_GET_ARY(p->stream, kAudioStreamPropertyAvailablePhysicalFormats,
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&formats, &n_formats);
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CHECK_CA_ERROR("could not get number of stream formats");
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for (int j = 0; j < n_formats; j++) {
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AudioStreamBasicDescription *stream_asbd = &formats[j].mFormat;
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ca_print_asbd(ao, "- ", stream_asbd);
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if (!out_fmt->mFormatID || ca_asbd_is_better(&asbd, out_fmt, stream_asbd))
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*out_fmt = *stream_asbd;
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}
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talloc_free(formats);
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if (!out_fmt->mFormatID) {
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MP_ERR(ao, "no format found\n");
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return -1;
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}
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return 0;
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coreaudio_error:
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return -1;
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}
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static int init(struct ao *ao)
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{
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struct priv *p = ao->priv;
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OSStatus err = ca_select_device(ao, ao->device, &p->device);
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CHECK_CA_ERROR_L(coreaudio_error_nounlock, "failed to select device");
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ao->format = af_fmt_from_planar(ao->format);
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if (!af_fmt_is_pcm(ao->format) && !af_fmt_is_spdif(ao->format)) {
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MP_ERR(ao, "Unsupported format.\n");
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goto coreaudio_error_nounlock;
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}
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uint32_t is_alive = 1;
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err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
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CHECK_CA_WARN("could not check whether device is alive");
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if (!is_alive)
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MP_WARN(ao , "device is not alive\n");
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err = ca_lock_device(p->device, &p->hog_pid);
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CHECK_CA_WARN("failed to set hogmode");
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err = ca_disable_mixing(ao, p->device, &p->changed_mixing);
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CHECK_CA_WARN("failed to disable mixing");
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if (select_stream(ao) < 0)
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goto coreaudio_error;
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AudioStreamBasicDescription hwfmt;
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if (find_best_format(ao, &hwfmt) < 0)
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goto coreaudio_error;
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err = CA_GET(p->stream, kAudioStreamPropertyPhysicalFormat,
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&p->original_asbd);
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CHECK_CA_ERROR("could not get stream's original physical format");
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// Even if changing the physical format fails, we can try using the current
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// virtual format.
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ca_change_physical_format_sync(ao, p->stream, hwfmt);
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if (!ca_init_chmap(ao, p->device))
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goto coreaudio_error;
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err = CA_GET(p->stream, kAudioStreamPropertyVirtualFormat, &p->stream_asbd);
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CHECK_CA_ERROR("could not get stream's virtual format");
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ca_print_asbd(ao, "virtual format", &p->stream_asbd);
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if (p->stream_asbd.mChannelsPerFrame > MP_NUM_CHANNELS) {
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MP_ERR(ao, "unsupported number of channels: %d > %d.\n",
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p->stream_asbd.mChannelsPerFrame, MP_NUM_CHANNELS);
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goto coreaudio_error;
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}
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int new_format = ca_asbd_to_mp_format(&p->stream_asbd);
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// If both old and new formats are spdif, avoid changing it due to the
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// imperfect mapping between mp and CA formats.
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if (!(af_fmt_is_spdif(ao->format) && af_fmt_is_spdif(new_format)))
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ao->format = new_format;
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if (!ao->format || af_fmt_is_planar(ao->format)) {
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MP_ERR(ao, "hardware format not supported\n");
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goto coreaudio_error;
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}
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ao->samplerate = p->stream_asbd.mSampleRate;
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if (ao->channels.num != p->stream_asbd.mChannelsPerFrame) {
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ca_get_active_chmap(ao, p->device, p->stream_asbd.mChannelsPerFrame,
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&ao->channels);
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}
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if (!ao->channels.num) {
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MP_ERR(ao, "number of channels changed, and unknown channel layout!\n");
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goto coreaudio_error;
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}
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p->hw_latency_us = ca_get_device_latency_us(ao, p->device);
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MP_VERBOSE(ao, "base latency: %d microseconds\n", (int)p->hw_latency_us);
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err = enable_property_listener(ao, true);
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CHECK_CA_ERROR("cannot install format change listener during init");
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err = AudioDeviceCreateIOProcID(p->device,
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(AudioDeviceIOProc)render_cb_compressed,
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(void *)ao,
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&p->render_cb);
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CHECK_CA_ERROR("failed to register audio render callback");
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return CONTROL_TRUE;
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coreaudio_error:
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err = enable_property_listener(ao, false);
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CHECK_CA_WARN("can't remove format change listener");
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err = ca_unlock_device(p->device, &p->hog_pid);
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CHECK_CA_WARN("can't release hog mode");
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coreaudio_error_nounlock:
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return CONTROL_ERROR;
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}
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static void uninit(struct ao *ao)
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{
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struct priv *p = ao->priv;
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OSStatus err = noErr;
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err = enable_property_listener(ao, false);
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CHECK_CA_WARN("can't remove device listener, this may cause a crash");
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err = AudioDeviceStop(p->device, p->render_cb);
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CHECK_CA_WARN("failed to stop audio device");
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err = AudioDeviceDestroyIOProcID(p->device, p->render_cb);
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CHECK_CA_WARN("failed to remove device render callback");
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if (!ca_change_physical_format_sync(ao, p->stream, p->original_asbd))
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MP_WARN(ao, "can't revert to original device format\n");
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err = ca_enable_mixing(ao, p->device, p->changed_mixing);
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CHECK_CA_WARN("can't re-enable mixing");
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err = ca_unlock_device(p->device, &p->hog_pid);
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CHECK_CA_WARN("can't release hog mode");
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}
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static void audio_pause(struct ao *ao)
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{
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struct priv *p = ao->priv;
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OSStatus err = AudioDeviceStop(p->device, p->render_cb);
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CHECK_CA_WARN("can't stop audio device");
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}
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static void audio_resume(struct ao *ao)
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{
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struct priv *p = ao->priv;
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OSStatus err = AudioDeviceStart(p->device, p->render_cb);
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CHECK_CA_WARN("can't start audio device");
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}
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#define OPT_BASE_STRUCT struct priv
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const struct ao_driver audio_out_coreaudio_exclusive = {
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.description = "CoreAudio Exclusive Mode",
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.name = "coreaudio_exclusive",
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.uninit = uninit,
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.init = init,
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.reset = audio_pause,
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.resume = audio_resume,
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.list_devs = ca_get_device_list,
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.priv_size = sizeof(struct priv),
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.priv_defaults = &(const struct priv){
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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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};
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