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authorwm4 <wm4@nowhere>2013-11-10 23:38:18 +0100
committerwm4 <wm4@nowhere>2013-11-12 23:29:53 +0100
commit347a86198b214b5e79b45d198c5cd2cc3c3a759a (patch)
treea8adf255ac2cedfa4bb29384e02243550c7a6647 /audio/decode
parentd1ee9ea261798f94fe958578d9748a69864c3058 (diff)
downloadmpv-347a86198b214b5e79b45d198c5cd2cc3c3a759a.tar.bz2
mpv-347a86198b214b5e79b45d198c5cd2cc3c3a759a.tar.xz
audio: switch output to mp_audio_buffer
Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
Diffstat (limited to 'audio/decode')
-rw-r--r--audio/decode/dec_audio.c54
-rw-r--r--audio/decode/dec_audio.h7
2 files changed, 22 insertions, 39 deletions
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index ef7993c83a..3f92c3c4e6 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -38,6 +38,7 @@
#include "dec_audio.h"
#include "ad.h"
#include "audio/format.h"
+#include "audio/audio_buffer.h"
#include "audio/filter/af.h"
@@ -230,18 +231,9 @@ int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
return 1;
}
-static void set_min_out_buffer_size(struct bstr *outbuf, int len)
-{
- size_t oldlen = talloc_get_size(outbuf->start);
- if (oldlen < len) {
- assert(outbuf->start); // talloc context should be already set
- mp_msg(MSGT_DECAUDIO, MSGL_V, "Increasing filtered audio buffer size "
- "from %zd to %d\n", oldlen, len);
- outbuf->start = talloc_realloc_size(NULL, outbuf->start, len);
- }
-}
-
-static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
+// Filter len bytes of input, put result into outbuf.
+static int filter_n_bytes(sh_audio_t *sh, struct mp_audio_buffer *outbuf,
+ int len)
{
assert(len - 1 + sh->audio_out_minsize <= sh->a_buffer_size);
@@ -280,10 +272,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
if (!filter_output)
return -1;
- int outlen = filter_output->samples * filter_output->sstride;
- set_min_out_buffer_size(outbuf, outbuf->len + outlen);
- memcpy(outbuf->start + outbuf->len, filter_output->planes[0], outlen);
- outbuf->len += outlen;
+ mp_audio_buffer_append(outbuf, filter_output);
// remove processed data from decoder buffer:
sh->a_buffer_len -= len;
@@ -292,20 +281,20 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
return error;
}
-/* Try to get at least minlen decoded+filtered bytes in outbuf
+/* Try to get at least minsamples decoded+filtered samples in outbuf
* (total length including possible existing data).
* Return 0 on success, -1 on error/EOF (not distinguished).
- * In the former case outbuf->len is always >= minlen on return.
- * In case of EOF/error it might or might not be.
- * Outbuf.start must be talloc-allocated, and will be reallocated
- * if needed to fit all filter output. */
-int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
+ * In the former case outbuf has at least minsamples buffered on return.
+ * In case of EOF/error it might or might not be. */
+int decode_audio(sh_audio_t *sh_audio, struct mp_audio_buffer *outbuf,
+ int minsamples)
{
// Indicates that a filter seems to be buffering large amounts of data
int huge_filter_buffer = 0;
+ int sstride =
+ af_fmt2bits(sh_audio->sample_format) / 8 * sh_audio->channels.num;
// Decoded audio must be cut at boundaries of this many bytes
- int bps = af_fmt2bits(sh_audio->sample_format) / 8;
- int unitsize = sh_audio->channels.num * bps * 16;
+ int unitsize = sstride * 16;
/* Filter output size will be about filter_multiplier times input size.
* If some filter buffers audio in big blocks this might only hold
@@ -322,9 +311,13 @@ int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
return -1;
max_decode_len -= max_decode_len % unitsize;
- while (minlen >= 0 && outbuf->len < minlen) {
+ while (minsamples >= 0 && mp_audio_buffer_samples(outbuf) < minsamples) {
+ struct af_stream *afs = sh_audio->afilter;
+ int out_sstride = afs->output.sstride;
+ int declen = (minsamples - mp_audio_buffer_samples(outbuf))
+ * out_sstride / filter_multiplier;
// + some extra for possible filter buffering
- int declen = (minlen - outbuf->len) / filter_multiplier + (unitsize << 5);
+ declen += unitsize << 5;
if (huge_filter_buffer)
/* Some filter must be doing significant buffering if the estimated
* input length didn't produce enough output from filters.
@@ -349,15 +342,6 @@ int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
return 0;
}
-void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte)
-{
- set_min_out_buffer_size(outbuf, outbuf->len + count);
- memmove(outbuf->start + count, outbuf->start, outbuf->len);
- memset(outbuf->start, byte, count);
- outbuf->len += count;
-}
-
-
void resync_audio_stream(sh_audio_t *sh_audio)
{
sh_audio->pts = MP_NOPTS_VALUE;
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
index b46f4282fb..3ec5954471 100644
--- a/audio/decode/dec_audio.h
+++ b/audio/decode/dec_audio.h
@@ -22,15 +22,14 @@
#include "audio/chmap.h"
#include "demux/stheader.h"
-struct bstr;
+struct mp_audio_buffer;
struct mp_decoder_list;
struct mp_decoder_list *mp_audio_decoder_list(void);
int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders);
-int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen);
-void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte);
+int decode_audio(sh_audio_t *sh_audio, struct mp_audio_buffer *outbuf,
+ int minsamples);
void resync_audio_stream(sh_audio_t *sh_audio);
-void skip_audio_frame(sh_audio_t *sh_audio);
void uninit_audio(sh_audio_t *sh_audio);
int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,