зеркало из https://github.com/mozilla/pjs.git
710 строки
21 KiB
C
710 строки
21 KiB
C
/* ***** BEGIN LICENSE BLOCK *****
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* Version: MPL 1.1/GPL 2.0/LGPL 2.1
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*
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* The contents of this file are subject to the Mozilla Public License Version
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* 1.1 (the "License"); you may not use this file except in compliance with
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* the License. You may obtain a copy of the License at
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* http://www.mozilla.org/MPL/
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*
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* Software distributed under the License is distributed on an "AS IS" basis,
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* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
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* for the specific language governing rights and limitations under the
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* License.
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*
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* The Initial Developer of the Original Code is
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* CSIRO
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* Portions created by the Initial Developer are Copyright (C) 2007
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* the Initial Developer. All Rights Reserved.
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*
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* Contributor(s): Michael Martin
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* Chris Double (chris.double@double.co.nz)
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*
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* Alternatively, the contents of this file may be used under the terms of
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* either the GNU General Public License Version 2 or later (the "GPL"), or
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* the GNU Lesser General Public License Version 2.1 or later (the "LGPL"),
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* in which case the provisions of the GPL or the LGPL are applicable instead
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* of those above. If you wish to allow use of your version of this file only
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* under the terms of either the GPL or the LGPL, and not to allow others to
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* use your version of this file under the terms of the MPL, indicate your
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* decision by deleting the provisions above and replace them with the notice
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* and other provisions required by the GPL or the LGPL. If you do not delete
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* the provisions above, a recipient may use your version of this file under
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* the terms of any one of the MPL, the GPL or the LGPL.
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*
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* ***** END LICENSE BLOCK ***** *
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*/
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#include <stdlib.h>
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#include <pthread.h>
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#include <alsa/asoundlib.h>
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#include "sydney_audio.h"
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/* ALSA implementation based heavily on sydney_audio_mac.c */
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/*
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* The audio interface is based on a "pull" I/O model, which means you
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* can't just provide a data buffer and tell the audio device to play; you must
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* register a callback and provide data as the device asks for it. To support
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* sydney audio's "write-to-play" style interface, we have to buffer up the
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* data as it arrives and feed it to the callback as required.
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*
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* This is handled by a simple linked list of buffers; data is always written
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* to the tail and read from the head. Each buffer tracks the start and end
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* positions of its contained data. Buffers are allocated when the tail buffer
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* fills, and freed when the head buffer empties. There is always at least one
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* buffer allocated.
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*
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* s e s e s e + data read
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* +++##### -> ######## -> ####---- # data written
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* ^ ^ - empty
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* bl_head bl_tail
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*/
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typedef struct sa_buf sa_buf;
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struct sa_buf {
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unsigned int size;
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unsigned int start;
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unsigned int end;
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sa_buf * next;
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unsigned char data[0];
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};
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struct sa_stream {
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snd_pcm_t* output_unit;
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pthread_t thread_id;
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pthread_mutex_t mutex;
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char playing;
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int64_t bytes_played;
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/* audio format info */
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unsigned int rate;
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unsigned int n_channels;
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unsigned int bytes_per_ch;
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/* buffer list */
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sa_buf * bl_head;
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sa_buf * bl_tail;
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int n_bufs;
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};
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/*
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* Use a default buffer size with enough room for one second of audio,
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* assuming stereo data at 44.1kHz with 32 bits per channel, and impose
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* a generous limit on the number of buffers.
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*/
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#define BUF_SIZE (2 * 44100 * 4)
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#define BUF_LIMIT 5
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#if BUF_LIMIT < 2
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#error BUF_LIMIT must be at least 2!
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#endif
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static void audio_callback(void* s);
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static sa_buf *new_buffer(void);
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/*
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* -----------------------------------------------------------------------------
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* Startup and shutdown functions
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* -----------------------------------------------------------------------------
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*/
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int
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sa_stream_create_pcm(
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sa_stream_t ** _s,
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const char * client_name,
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sa_mode_t mode,
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sa_pcm_format_t format,
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unsigned int rate,
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unsigned int n_channels
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) {
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sa_stream_t * s = 0;
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/*
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* Make sure we return a NULL stream pointer on failure.
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*/
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if (_s == NULL) {
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return SA_ERROR_INVALID;
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}
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*_s = NULL;
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if (mode != SA_MODE_WRONLY) {
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return SA_ERROR_NOT_SUPPORTED;
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}
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if (format != SA_PCM_FORMAT_S16_LE) {
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return SA_ERROR_NOT_SUPPORTED;
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}
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/*
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* Allocate the instance and required resources.
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*/
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if ((s = malloc(sizeof(sa_stream_t))) == NULL) {
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return SA_ERROR_OOM;
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}
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if ((s->bl_head = new_buffer()) == NULL) {
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free(s);
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return SA_ERROR_OOM;
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}
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if (pthread_mutex_init(&s->mutex, NULL) != 0) {
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free(s->bl_head);
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free(s);
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return SA_ERROR_SYSTEM;
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}
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s->output_unit = NULL;
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s->thread_id = 0;
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s->playing = 0;
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s->bytes_played = 0;
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s->rate = rate;
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s->n_channels = n_channels;
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s->bytes_per_ch = 2;
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s->bl_tail = s->bl_head;
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s->n_bufs = 1;
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*_s = s;
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return SA_SUCCESS;
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}
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int
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sa_stream_open(sa_stream_t *s) {
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if (s == NULL) {
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return SA_ERROR_NO_INIT;
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}
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if (s->output_unit != NULL) {
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return SA_ERROR_INVALID;
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}
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if (snd_pcm_open(&s->output_unit,
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"default",
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SND_PCM_STREAM_PLAYBACK,
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0) < 0) {
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return SA_ERROR_NO_DEVICE;
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}
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if (snd_pcm_set_params(s->output_unit,
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SND_PCM_FORMAT_S16_LE,
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SND_PCM_ACCESS_RW_INTERLEAVED,
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s->n_channels,
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s->rate,
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1,
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0) < 0) {
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snd_pcm_close(s->output_unit);
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s->output_unit = NULL;
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return SA_ERROR_NOT_SUPPORTED;
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}
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return SA_SUCCESS;
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}
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int
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sa_stream_destroy(sa_stream_t *s) {
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int result = SA_SUCCESS;
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if (s == NULL) {
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return SA_SUCCESS;
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}
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pthread_mutex_lock(&s->mutex);
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/*
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* This causes the thread sending data to ALSA to stop
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*/
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s->thread_id = 0;
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/*
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* Shut down the audio output device.
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*/
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if (s->output_unit != NULL) {
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if (s->playing && snd_pcm_close(s->output_unit) < 0) {
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result = SA_ERROR_SYSTEM;
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}
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}
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pthread_mutex_unlock(&s->mutex);
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/*
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* Release resources.
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*/
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if (pthread_mutex_destroy(&s->mutex) != 0) {
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result = SA_ERROR_SYSTEM;
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}
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while (s->bl_head != NULL) {
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sa_buf * next = s->bl_head->next;
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free(s->bl_head);
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s->bl_head = next;
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}
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free(s);
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return result;
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}
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/*
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* -----------------------------------------------------------------------------
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* Data read and write functions
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* -----------------------------------------------------------------------------
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*/
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int
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sa_stream_write(sa_stream_t *s, const void *data, size_t nbytes) {
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int result = SA_SUCCESS;
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if (s == NULL || s->output_unit == NULL) {
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return SA_ERROR_NO_INIT;
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}
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if (nbytes == 0) {
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return SA_SUCCESS;
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}
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pthread_mutex_lock(&s->mutex);
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/*
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* Append the new data to the end of our buffer list.
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*/
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while (1) {
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unsigned int avail = s->bl_tail->size - s->bl_tail->end;
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if (nbytes <= avail) {
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/*
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* The new data will fit into the current tail buffer, so
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* just copy it in and we're done.
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*/
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memcpy(s->bl_tail->data + s->bl_tail->end, data, nbytes);
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s->bl_tail->end += nbytes;
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break;
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} else {
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/*
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* Copy what we can into the tail and allocate a new buffer
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* for the rest.
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*/
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memcpy(s->bl_tail->data + s->bl_tail->end, data, avail);
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s->bl_tail->end += avail;
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data = ((unsigned char *)data) + avail;
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nbytes -= avail;
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/*
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* If we still have data left to copy but we've hit the limit of
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* allowable buffer allocations, we need to spin for a bit to allow
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* the audio callback function to slurp some more data up.
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*/
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if (nbytes > 0 && s->n_bufs == BUF_LIMIT) {
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#ifdef TIMING_TRACE
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printf("#"); /* too much audio data */
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#endif
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if (!s->playing) {
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/*
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* We haven't even started playing yet! That means the
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* BUF_SIZE/BUF_LIMIT values are too low... Not much we can
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* do here; spinning won't help because the audio callback
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* hasn't been enabled yet. Oh well, error time.
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*/
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printf("Too much audio data received before audio device enabled!\n");
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result = SA_ERROR_SYSTEM;
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break;
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}
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while (s->n_bufs == BUF_LIMIT) {
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struct timespec ts = {0, 1000000};
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pthread_mutex_unlock(&s->mutex);
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nanosleep(&ts, NULL);
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pthread_mutex_lock(&s->mutex);
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}
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}
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/*
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* Allocate a new tail buffer, and go 'round again to fill it up.
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*/
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if ((s->bl_tail->next = new_buffer()) == NULL) {
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result = SA_ERROR_OOM;
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break;
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}
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s->n_bufs++;
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s->bl_tail = s->bl_tail->next;
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} /* if (nbytes <= avail), else */
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} /* while (1) */
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pthread_mutex_unlock(&s->mutex);
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/*
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* Once we have our first block of audio data, enable the audio callback
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* function. This doesn't need to be protected by the mutex, because
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* s->playing is not used in the audio callback thread, and it's probably
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* better not to be inside the lock when we enable the audio callback.
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*/
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if (!s->playing) {
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s->playing = 1;
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if (pthread_create(&s->thread_id, NULL, (void *)audio_callback, s) != 0) {
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result = SA_ERROR_SYSTEM;
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}
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}
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return result;
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}
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static void audio_callback(void* data)
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{
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sa_stream_t* s = (sa_stream_t*)data;
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snd_pcm_uframes_t buffer_size;
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snd_pcm_uframes_t period_size;
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unsigned int bytes_per_frame = s->n_channels * s->bytes_per_ch;
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char* buffer = 0;
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#ifdef TIMING_TRACE
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printf("."); /* audio read 'tick' */
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#endif
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snd_pcm_get_params(s->output_unit, &buffer_size, &period_size);
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buffer = malloc(period_size * bytes_per_frame);
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while(1) {
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char* dst = buffer;
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unsigned int bytes_to_copy = period_size * bytes_per_frame;
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snd_pcm_sframes_t frames;
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pthread_mutex_lock(&s->mutex);
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if (!s->thread_id)
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break;
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/*
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* Consume data from the start of the buffer list.
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*/
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while (1) {
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unsigned int avail = s->bl_head->end - s->bl_head->start;
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assert(s->bl_head->start <= s->bl_head->end);
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if (avail >= bytes_to_copy) {
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/*
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* We have all we need in the head buffer, so just grab it and go.
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*/
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memcpy(dst, s->bl_head->data + s->bl_head->start, bytes_to_copy);
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s->bl_head->start += bytes_to_copy;
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s->bytes_played += bytes_to_copy;
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break;
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} else {
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sa_buf* next = 0;
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/*
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* Copy what we can from the head and move on to the next buffer.
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*/
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memcpy(dst, s->bl_head->data + s->bl_head->start, avail);
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s->bl_head->start += avail;
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dst += avail;
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bytes_to_copy -= avail;
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s->bytes_played += avail;
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/*
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* We want to free the now-empty buffer, but not if it's also the
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* current tail. If it is the tail, we don't have enough data to fill
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* the destination buffer, so we'll just zero it out and give up.
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*/
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next = s->bl_head->next;
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if (next == NULL) {
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#ifdef TIMING_TRACE
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printf("!"); /* not enough audio data */
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#endif
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memset(dst, 0, bytes_to_copy);
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break;
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}
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free(s->bl_head);
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s->bl_head = next;
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s->n_bufs--;
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} /* if (avail >= bytes_to_copy), else */
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} /* while (1) */
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pthread_mutex_unlock(&s->mutex);
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frames = snd_pcm_writei(s->output_unit, buffer, period_size);
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if (frames < 0) {
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frames = snd_pcm_recover(s->output_unit, frames, 1);
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if (frames < 0) {
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printf("snc_pcm_recover error: %s\n", snd_strerror(frames));
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}
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if(frames > 0 && frames < period_size)
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printf("short write (expected %d, wrote %d)\n", (int)period_size, (int)frames);;
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}
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}
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free(buffer);
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}
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/*
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* -----------------------------------------------------------------------------
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* General query and support functions
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* -----------------------------------------------------------------------------
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*/
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int
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sa_stream_get_write_size(sa_stream_t *s, size_t *size) {
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sa_buf * b;
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size_t used = 0;
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if (s == NULL || s->output_unit == NULL) {
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return SA_ERROR_NO_INIT;
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}
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pthread_mutex_lock(&s->mutex);
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/*
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* Sum up the used portions of our buffers and subtract that from
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* the pre-defined max allowed allocation.
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*/
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for (b = s->bl_head; b != NULL; b = b->next) {
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used += b->end - b->start;
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}
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*size = BUF_SIZE * BUF_LIMIT - used;
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pthread_mutex_unlock(&s->mutex);
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return SA_SUCCESS;
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}
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int
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sa_stream_get_position(sa_stream_t *s, sa_position_t position, int64_t *pos) {
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if (s == NULL || s->output_unit == NULL) {
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return SA_ERROR_NO_INIT;
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}
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if (position != SA_POSITION_WRITE_SOFTWARE) {
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return SA_ERROR_NOT_SUPPORTED;
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}
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pthread_mutex_lock(&s->mutex);
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*pos = s->bytes_played;
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pthread_mutex_unlock(&s->mutex);
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return SA_SUCCESS;
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}
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int
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sa_stream_pause(sa_stream_t *s) {
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if (s == NULL || s->output_unit == NULL) {
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return SA_ERROR_NO_INIT;
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}
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pthread_mutex_lock(&s->mutex);
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#if 0 /* TODO */
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AudioOutputUnitStop(s->output_unit);
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#endif
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pthread_mutex_unlock(&s->mutex);
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return SA_SUCCESS;
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}
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int
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sa_stream_resume(sa_stream_t *s) {
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if (s == NULL || s->output_unit == NULL) {
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return SA_ERROR_NO_INIT;
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}
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pthread_mutex_lock(&s->mutex);
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/*
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* The audio device resets its mSampleTime counter after pausing,
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* so we need to clear our tracking value to keep that in sync.
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*/
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s->bytes_played = 0;
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#if 0 /* TODO */
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AudioOutputUnitStart(s->output_unit);
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#endif
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pthread_mutex_unlock(&s->mutex);
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return SA_SUCCESS;
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}
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static sa_buf *
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new_buffer(void) {
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sa_buf * b = malloc(sizeof(sa_buf) + BUF_SIZE);
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if (b != NULL) {
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b->size = BUF_SIZE;
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b->start = 0;
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b->end = 0;
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b->next = NULL;
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}
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return b;
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}
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/*
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* -----------------------------------------------------------------------------
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* Extension functions
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* -----------------------------------------------------------------------------
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*/
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int
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sa_stream_set_volume_abs(sa_stream_t *s, float vol) {
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snd_mixer_t* mixer = 0;
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snd_mixer_elem_t* elem = 0;
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if (s == NULL || s->output_unit == NULL) {
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return SA_ERROR_NO_INIT;
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}
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if (snd_mixer_open(&mixer, 0) < 0) {
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return SA_ERROR_SYSTEM;
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}
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if (snd_mixer_attach(mixer, "default") < 0) {
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snd_mixer_close(mixer);
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return SA_ERROR_SYSTEM;
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}
|
|
|
|
if (snd_mixer_selem_register(mixer, NULL, NULL) < 0) {
|
|
snd_mixer_close(mixer);
|
|
return SA_ERROR_SYSTEM;
|
|
}
|
|
|
|
if (snd_mixer_load(mixer) < 0) {
|
|
snd_mixer_close(mixer);
|
|
return SA_ERROR_SYSTEM;
|
|
}
|
|
|
|
#if 0
|
|
snd_mixer_elem_t* elem = 0;
|
|
for (elem = snd_mixer_first_elem(mixer); elem != NULL; elem = snd_mixer_elem_next(elem)) {
|
|
if (snd_mixer_selem_has_playback_volume(elem)) {
|
|
printf("Playback %s\n", snd_mixer_selem_get_name(elem));
|
|
}
|
|
else {
|
|
printf("No Playback: %s\n", snd_mixer_selem_get_name(elem));
|
|
}
|
|
}
|
|
#endif
|
|
elem = snd_mixer_first_elem(mixer);
|
|
if (elem && snd_mixer_selem_has_playback_volume(elem)) {
|
|
long min = 0;
|
|
long max = 0;
|
|
if (snd_mixer_selem_get_playback_volume_range(elem, &min, &max) >= 0) {
|
|
snd_mixer_selem_set_playback_volume_all(elem, (max-min)*vol + min);
|
|
}
|
|
}
|
|
snd_mixer_close(mixer);
|
|
|
|
return SA_SUCCESS;
|
|
}
|
|
|
|
|
|
int
|
|
sa_stream_get_volume_abs(sa_stream_t *s, float *vol) {
|
|
snd_mixer_t* mixer = 0;
|
|
snd_mixer_elem_t* elem = 0;
|
|
long value = 0;
|
|
|
|
if (s == NULL || s->output_unit == NULL) {
|
|
return SA_ERROR_NO_INIT;
|
|
}
|
|
|
|
if (snd_mixer_open(&mixer, 0) < 0) {
|
|
return SA_ERROR_SYSTEM;
|
|
}
|
|
|
|
if (snd_mixer_attach(mixer, "default") < 0) {
|
|
snd_mixer_close(mixer);
|
|
return SA_ERROR_SYSTEM;
|
|
}
|
|
|
|
if (snd_mixer_selem_register(mixer, NULL, NULL) < 0) {
|
|
snd_mixer_close(mixer);
|
|
return SA_ERROR_SYSTEM;
|
|
}
|
|
|
|
if (snd_mixer_load(mixer) < 0) {
|
|
snd_mixer_close(mixer);
|
|
return SA_ERROR_SYSTEM;
|
|
}
|
|
|
|
elem = snd_mixer_first_elem(mixer);
|
|
if (elem && snd_mixer_selem_get_playback_volume(elem, 0, &value) >= 0) {
|
|
long min = 0;
|
|
long max = 0;
|
|
if (snd_mixer_selem_get_playback_volume_range(elem, &min, &max) >= 0) {
|
|
*vol = (float)(value-min)/(float)(max-min);
|
|
}
|
|
}
|
|
snd_mixer_close(mixer);
|
|
|
|
return SA_SUCCESS;
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
* -----------------------------------------------------------------------------
|
|
* Unsupported functions
|
|
* -----------------------------------------------------------------------------
|
|
*/
|
|
#define UNSUPPORTED(func) func { return SA_ERROR_NOT_SUPPORTED; }
|
|
|
|
UNSUPPORTED(int sa_stream_create_opaque(sa_stream_t **s, const char *client_name, sa_mode_t mode, const char *codec))
|
|
UNSUPPORTED(int sa_stream_set_write_lower_watermark(sa_stream_t *s, size_t size))
|
|
UNSUPPORTED(int sa_stream_set_read_lower_watermark(sa_stream_t *s, size_t size))
|
|
UNSUPPORTED(int sa_stream_set_write_upper_watermark(sa_stream_t *s, size_t size))
|
|
UNSUPPORTED(int sa_stream_set_read_upper_watermark(sa_stream_t *s, size_t size))
|
|
UNSUPPORTED(int sa_stream_set_channel_map(sa_stream_t *s, const sa_channel_t map[], unsigned int n))
|
|
UNSUPPORTED(int sa_stream_set_xrun_mode(sa_stream_t *s, sa_xrun_mode_t mode))
|
|
UNSUPPORTED(int sa_stream_set_non_interleaved(sa_stream_t *s, int enable))
|
|
UNSUPPORTED(int sa_stream_set_dynamic_rate(sa_stream_t *s, int enable))
|
|
UNSUPPORTED(int sa_stream_set_driver(sa_stream_t *s, const char *driver))
|
|
UNSUPPORTED(int sa_stream_start_thread(sa_stream_t *s, sa_event_callback_t callback))
|
|
UNSUPPORTED(int sa_stream_stop_thread(sa_stream_t *s))
|
|
UNSUPPORTED(int sa_stream_change_device(sa_stream_t *s, const char *device_name))
|
|
UNSUPPORTED(int sa_stream_change_read_volume(sa_stream_t *s, const int32_t vol[], unsigned int n))
|
|
UNSUPPORTED(int sa_stream_change_write_volume(sa_stream_t *s, const int32_t vol[], unsigned int n))
|
|
UNSUPPORTED(int sa_stream_change_rate(sa_stream_t *s, unsigned int rate))
|
|
UNSUPPORTED(int sa_stream_change_meta_data(sa_stream_t *s, const char *name, const void *data, size_t size))
|
|
UNSUPPORTED(int sa_stream_change_user_data(sa_stream_t *s, const void *value))
|
|
UNSUPPORTED(int sa_stream_set_adjust_rate(sa_stream_t *s, sa_adjust_t direction))
|
|
UNSUPPORTED(int sa_stream_set_adjust_nchannels(sa_stream_t *s, sa_adjust_t direction))
|
|
UNSUPPORTED(int sa_stream_set_adjust_pcm_format(sa_stream_t *s, sa_adjust_t direction))
|
|
UNSUPPORTED(int sa_stream_set_adjust_watermarks(sa_stream_t *s, sa_adjust_t direction))
|
|
UNSUPPORTED(int sa_stream_get_mode(sa_stream_t *s, sa_mode_t *access_mode))
|
|
UNSUPPORTED(int sa_stream_get_codec(sa_stream_t *s, char *codec, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_pcm_format(sa_stream_t *s, sa_pcm_format_t *format))
|
|
UNSUPPORTED(int sa_stream_get_rate(sa_stream_t *s, unsigned int *rate))
|
|
UNSUPPORTED(int sa_stream_get_nchannels(sa_stream_t *s, int *nchannels))
|
|
UNSUPPORTED(int sa_stream_get_user_data(sa_stream_t *s, void **value))
|
|
UNSUPPORTED(int sa_stream_get_write_lower_watermark(sa_stream_t *s, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_read_lower_watermark(sa_stream_t *s, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_write_upper_watermark(sa_stream_t *s, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_read_upper_watermark(sa_stream_t *s, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_channel_map(sa_stream_t *s, sa_channel_t map[], unsigned int *n))
|
|
UNSUPPORTED(int sa_stream_get_xrun_mode(sa_stream_t *s, sa_xrun_mode_t *mode))
|
|
UNSUPPORTED(int sa_stream_get_non_interleaved(sa_stream_t *s, int *enabled))
|
|
UNSUPPORTED(int sa_stream_get_dynamic_rate(sa_stream_t *s, int *enabled))
|
|
UNSUPPORTED(int sa_stream_get_driver(sa_stream_t *s, char *driver_name, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_device(sa_stream_t *s, char *device_name, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_read_volume(sa_stream_t *s, int32_t vol[], unsigned int *n))
|
|
UNSUPPORTED(int sa_stream_get_write_volume(sa_stream_t *s, int32_t vol[], unsigned int *n))
|
|
UNSUPPORTED(int sa_stream_get_meta_data(sa_stream_t *s, const char *name, void*data, size_t *size))
|
|
UNSUPPORTED(int sa_stream_get_adjust_rate(sa_stream_t *s, sa_adjust_t *direction))
|
|
UNSUPPORTED(int sa_stream_get_adjust_nchannels(sa_stream_t *s, sa_adjust_t *direction))
|
|
UNSUPPORTED(int sa_stream_get_adjust_pcm_format(sa_stream_t *s, sa_adjust_t *direction))
|
|
UNSUPPORTED(int sa_stream_get_adjust_watermarks(sa_stream_t *s, sa_adjust_t *direction))
|
|
UNSUPPORTED(int sa_stream_get_state(sa_stream_t *s, sa_state_t *state))
|
|
UNSUPPORTED(int sa_stream_get_event_error(sa_stream_t *s, sa_error_t *error))
|
|
UNSUPPORTED(int sa_stream_get_event_notify(sa_stream_t *s, sa_notify_t *notify))
|
|
UNSUPPORTED(int sa_stream_read(sa_stream_t *s, void *data, size_t nbytes))
|
|
UNSUPPORTED(int sa_stream_read_ni(sa_stream_t *s, unsigned int channel, void *data, size_t nbytes))
|
|
UNSUPPORTED(int sa_stream_write_ni(sa_stream_t *s, unsigned int channel, const void *data, size_t nbytes))
|
|
UNSUPPORTED(int sa_stream_pwrite(sa_stream_t *s, const void *data, size_t nbytes, int64_t offset, sa_seek_t whence))
|
|
UNSUPPORTED(int sa_stream_pwrite_ni(sa_stream_t *s, unsigned int channel, const void *data, size_t nbytes, int64_t offset, sa_seek_t whence))
|
|
UNSUPPORTED(int sa_stream_get_read_size(sa_stream_t *s, size_t *size))
|
|
UNSUPPORTED(int sa_stream_drain(sa_stream_t *s))
|
|
|
|
const char *sa_strerror(int code) { return NULL; }
|
|
|