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MPEGPlyaer: A bit of audio mutation. Remove a useless thread state. Take some control over the buffer away from the audio thread itself. Some atomicity corrections.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@28984 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Sevakis 2011-01-07 05:17:28 +00:00
parent b1a1831a17
commit 39107956ab
5 changed files with 141 additions and 84 deletions
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57
apps/plugins/mpegplayer/audio_thread.c
View file

@ -112,7 +112,7 @@ static inline void audiodesc_queue_add_tail(void)
audio_queue.write++;
}
/* Increments the queue tail position - leaves one slot as current */
/* Increments the queue head position - leaves one slot as current */
static inline bool audiodesc_queue_remove_head(void)
{
if (audio_queue.write == audio_queue.read)
@ -375,8 +375,6 @@ static void audio_thread_msg(struct audio_thread_data *td)
case TSTATE_INIT:
td->state = TSTATE_DECODE;
case TSTATE_DECODE:
case TSTATE_RENDER_WAIT:
case TSTATE_RENDER_WAIT_END:
break;
case TSTATE_EOS:
@ -455,7 +453,6 @@ static void audio_thread_msg(struct audio_thread_data *td)
{
case TSTATE_DECODE:
case TSTATE_RENDER_WAIT:
case TSTATE_RENDER_WAIT_END:
/* These return when in playing state */
return;
}
@ -512,7 +509,6 @@ static void audio_thread(void)
/* These states are the only ones that should return */
case TSTATE_DECODE: goto audio_decode;
case TSTATE_RENDER_WAIT: goto render_wait;
case TSTATE_RENDER_WAIT_END: goto render_wait_end;
/* Anything else is interpreted as an exit */
default:
{
@ -538,29 +534,28 @@ static void audio_thread(void)
case STREAM_DATA_END:
{
if (audio_queue.used > MAD_BUFFER_GUARD)
break;
break; /* Still have frames to decode */
/* Used up remainder of compressed audio buffer.
* Force any residue to play if audio ended before
* reaching the threshold */
td.state = TSTATE_RENDER_WAIT_END;
/* Used up remainder of compressed audio buffer. Wait for
* samples on PCM buffer to finish playing. */
audio_queue_reset();
render_wait_end:
pcm_output_drain();
while (pcm_output_used() > (ssize_t)PCMOUT_LOW_WM)
while (1)
{
if (pcm_output_empty())
{
td.state = TSTATE_EOS;
stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
break;
}
pcm_output_drain();
str_get_msg_w_tmo(&audio_str, &td.ev, 1);
if (td.ev.id != SYS_TIMEOUT)
goto message_process;
break;
}
td.state = TSTATE_EOS;
if (td.status == STREAM_PLAYING)
stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
rb->yield();
goto message_wait;
} /* STREAM_DATA_END: */
}
@ -606,11 +601,9 @@ static void audio_thread(void)
/* This is too hard - bail out */
td.state = TSTATE_EOS;
if (td.status == STREAM_PLAYING)
stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
td.status = STREAM_ERROR;
stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
goto message_wait;
}
@ -643,15 +636,15 @@ static void audio_thread(void)
render_wait:
if (synth.pcm.length > 0)
{
struct pcm_frame_header *dst_hdr = pcm_output_get_buffer();
const char *src[2] =
{ (char *)synth.pcm.samples[0], (char *)synth.pcm.samples[1] };
int out_count = (synth.pcm.length * CLOCK_RATE
+ (td.samplerate - 1)) / td.samplerate;
ssize_t size = sizeof(*dst_hdr) + out_count*4;
unsigned char *out_buf;
ssize_t size = out_count*4;
/* Wait for required amount of free buffer space */
while (pcm_output_free() < size)
while ((out_buf = pcm_output_get_buffer(&size)) == NULL)
{
/* Wait one frame */
int timeout = out_count*HZ / td.samplerate;
@ -660,21 +653,17 @@ static void audio_thread(void)
goto message_process;
}
out_count = rb->dsp_process(td.dsp, dst_hdr->data, src,
synth.pcm.length);
out_count = rb->dsp_process(td.dsp, out_buf, src, synth.pcm.length);
if (out_count <= 0)
break;
dst_hdr->size = sizeof(*dst_hdr) + out_count*4;
dst_hdr->time = audio_queue.curr->time;
/* Make this data available to DMA */
pcm_output_commit_data(out_count*4, audio_queue.curr->time);
/* As long as we're on this timestamp, the time is just
incremented by the number of samples */
audio_queue.curr->time += out_count;
/* Make this data available to DMA */
pcm_output_add_data();
}
rb->yield();