# define MACHINE_AO_FMT AO_FMT_LITTLE
#endif
-/** @brief How many seconds of input to buffer
+/** @brief Minimum number of frames to try to play at once
*
- * While any given connection has this much audio buffered, no more reads will
- * be issued for that connection. The decoder will have to wait.
- */
-#define BUFFER_SECONDS 5
-
-/** @brief Frame batch size
- *
- * This controls how many frames are written in one go.
+ * The main loop will only attempt to play any audio when this many
+ * frames are available (or the current track has reached the end).
+ * The actual number of frames it attempts to play will often be
+ * larger than this (up to three times).
*
* For ALSA we request a buffer of three times this size and set the low
* watermark to this amount. The goal is then to keep between 1 and 3 times
* this many frames in play.
*
- * For all backends we attempt to play up to three times this many frames per
+ * For other we attempt to play up to three times this many frames per
* shot. In practice we will often only send much less than this.
*/
#define FRAMES 4096
/** @brief Bytes to send per network packet
+ *
+ * This is the maximum number of bytes we pass to write(2); to determine actual
+ * packet sizes, add a UDP header and an IP header (and a link layer header if
+ * it's the link layer size you care about).
*
* Don't make this too big or arithmetic will start to overflow.
*/
-#define NETWORK_BYTES (1024+sizeof(struct rtp_header))
-
-/** @brief Maximum RTP playahead (ms) */
-#define RTP_AHEAD_MS 1000
+#define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
/** @brief Maximum number of FDs to poll for */
#define NFDS 256
* of these but rearranging the queue can cause there to be more.
*/
struct track {
- struct track *next; /* next track */
+ /** @brief Next track */
+ struct track *next;
+
+ /** @brief Input file descriptor */
int fd; /* input FD */
- char id[24]; /* ID */
- size_t start, used; /* start + bytes used */
- int eof; /* input is at EOF */
- int got_format; /* got format yet? */
- ao_sample_format format; /* sample format */
- unsigned long long played; /* number of frames played */
- char *buffer; /* sample buffer */
- size_t size; /* sample buffer size */
- int slot; /* poll array slot */
+
+ /** @brief Track ID */
+ char id[24];
+
+ /** @brief Start position of data in buffer */
+ size_t start;
+
+ /** @brief Number of bytes of data in buffer */
+ size_t used;
+
+ /** @brief Set @c fd is at EOF */
+ int eof;
+
+ /** @brief Total number of frames played */
+ unsigned long long played;
+
+ /** @brief Slot in @ref fds */
+ int slot;
+
+ /** @brief Set when playable
+ *
+ * A track becomes playable whenever it fills its buffer or reaches EOF; it
+ * stops being playable when it entirely empties its buffer. Tracks start
+ * out life not playable.
+ */
+ int playable;
+
+ /** @brief Input buffer
+ *
+ * 1Mbyte is enough for nearly 6s of 44100Hz 16-bit stereo
+ */
+ char buffer[1048576];
};
/** @brief Structure of a backend */
/** @brief Flags
*
- * Possible values
- * - @ref FIXED_FORMAT
+ * This field is currently not used and must be 0.
*/
unsigned flags;
-/** @brief Lock to configured sample format */
-#define FIXED_FORMAT 0x0001
/** @brief Initialization
*
*
* Called to activate the output device.
*
- * After this function succeeds, @ref ready should be non-0. As well as
- * opening the audio device, this function is responsible for reconfiguring
- * if it necessary to cope with different samples formats (for backends that
- * don't demand a single fixed sample format for the lifetime of the server).
+ * On input @ref device_state may be anything. If it is @ref
+ * device_open then the device is already open but might be using
+ * the wrong sample format. The device should be reconfigured to
+ * use the right sample format.
+ *
+ * If it is @ref device_error then a retry is underway and an
+ * attempt to recover or re-open the device (with the right sample
+ * format) should be made.
+ *
+ * If it is @ref device_closed then the device should be opened with
+ * the right sample format.
+ *
+ * Some devices are effectively always open and have no error state, in which
+ * case this callback can be NULL. Note that @ref device_state still
+ * switches between @ref device_open and @ref device_closed in this case.
*/
- int (*activate)(void);
+ void (*activate)(void);
/** @brief Play sound
* @param frames Number of frames to play
* @return Number of frames actually played
+ *
+ * If an error occurs (and it is not immediately recovered) this
+ * should set @ref device_state to @ref device_error.
*/
size_t (*play)(size_t frames);
/** @brief Deactivation
*
- * Called to deactivate the sound device. This is the inverse of
- * @c activate above.
+ * Called to deactivate the sound device. This is the inverse of @c
+ * activate above.
+ *
+ * For sound devices that are open all the time and have no error
+ * state, this callback can be NULL. Note that @ref device_state
+ * still switches between @ref device_open and @ref device_closed in
+ * this case.
*/
void (*deactivate)(void);
/** @brief Called before poll()
+ * @param timeoutp Pointer to timeout
+ *
+ * Called before the call to poll().
+ *
+ * If desirable, should call addfd() to update the FD array and stash the
+ * slot number somewhere safe. This will only be called if @ref device_state
+ * is @ref device_open.
*
- * Called before the call to poll(). Should call addfd() to update the FD
- * array and stash the slot number somewhere safe.
+ * @p timeoutp points to the poll timeout value in milliseconds. It may be
+ * reduced, but never increased.
+ *
+ * NB you can NOT assume that @c beforepoll is always called before @c play.
*/
- void (*beforepoll)(void);
+ void (*beforepoll)(int *timeoutp);
/** @brief Called after poll()
- * @return 0 if we could play, non-0 if not
+ * @return 1 if output device ready for play, 0 otherwise
*
- * Called after the call to poll(). Should arrange to play some audio if the
- * output device is ready.
+ * Called after the call to poll(). This will only be called if
+ * @ref device_state = @ref device_open.
*
- * The return value should be 0 if the device was ready to play, or nonzero
- * if it was not.
+ * The return value should be 1 if the device was ready to play, or
+ * 0 if it was not.
*/
- int (*afterpoll)(void);
+ int (*ready)(void);
};
-/** @brief Linked list of all prepared tracks */
-extern struct track *tracks;
+/** @brief Possible device states */
+enum device_states {
+ /** @brief The device is closed */
+ device_closed,
-/** @brief Playing track, or NULL */
+ /** @brief The device is open and ready to receive sound
+ *
+ * The current device sample format is potentially part of this state.
+ */
+ device_open,
+
+ /** @brief An error has occurred on the device
+ *
+ * This state is used to ensure that a small interval is left
+ * between retrying the device. If errors just set @ref
+ * device_closed then the main loop would busy-wait on broken output
+ * devices.
+ *
+ * The current device sample format is potentially part of this state.
+ */
+ device_error
+};
+
+extern enum device_states device_state;
+extern struct track *tracks;
extern struct track *playing;
+extern const struct speaker_backend network_backend;
+extern const struct speaker_backend alsa_backend;
+extern const struct speaker_backend command_backend;
+extern const struct speaker_backend coreaudio_backend;
+extern const struct speaker_backend oss_backend;
+
+extern struct pollfd fds[NFDS];
+extern int fdno;
+extern size_t bpf;
+extern int idled;
+
+int addfd(int fd, int events);
+void abandon(void);
+
#endif /* SPEAKER_H */
/*
~mdw / disorder / blobdiff