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-rw-r--r--audio/decode/ad.h4
-rw-r--r--audio/decode/ad_lavc.c33
-rw-r--r--audio/decode/ad_spdif.c41
-rw-r--r--audio/decode/dec_audio.c33
-rw-r--r--audio/decode/dec_audio.h7
5 files changed, 59 insertions, 59 deletions
diff --git a/audio/decode/ad.h b/audio/decode/ad.h
index 0af05e1827..a8384c277f 100644
--- a/audio/decode/ad.h
+++ b/audio/decode/ad.h
@@ -23,7 +23,7 @@
#include "demux/demux.h"
#include "audio/format.h"
-#include "audio/audio.h"
+#include "audio/aframe.h"
#include "dec_audio.h"
struct mp_decoder_list;
@@ -39,7 +39,7 @@ struct ad_functions {
bool (*send_packet)(struct dec_audio *da, struct demux_packet *pkt);
// Return whether decoding is still going on (false if EOF was reached).
// Never returns false & *out set, but can return true with !*out.
- bool (*receive_frame)(struct dec_audio *da, struct mp_audio **out);
+ bool (*receive_frame)(struct dec_audio *da, struct mp_aframe **out);
};
enum ad_ctrl {
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index d701630fc6..fb429d567b 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -40,7 +40,6 @@
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
- struct mp_audio frame;
bool force_channel_map;
uint32_t skip_samples, trim_samples;
bool preroll_done;
@@ -191,7 +190,7 @@ static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
return true;
}
-static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
+static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
@@ -217,25 +216,18 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
- struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
+ struct mp_aframe *mpframe = mp_aframe_from_avframe(priv->avframe);
if (!mpframe)
return true;
- struct mp_chmap lavc_chmap = mpframe->channels;
- if (lavc_chmap.num != avctx->channels)
- mp_chmap_from_channels(&lavc_chmap, avctx->channels);
- if (priv->force_channel_map) {
- if (lavc_chmap.num == da->codec->channels.num)
- lavc_chmap = da->codec->channels;
- }
- mp_audio_set_channels(mpframe, &lavc_chmap);
+ if (priv->force_channel_map)
+ mp_aframe_set_chmap(mpframe, &da->codec->channels);
- mpframe->pts = out_pts;
+ if (out_pts == MP_NOPTS_VALUE)
+ out_pts = priv->next_pts;
+ mp_aframe_set_pts(mpframe, out_pts);
- if (mpframe->pts == MP_NOPTS_VALUE)
- mpframe->pts = priv->next_pts;
- if (mpframe->pts != MP_NOPTS_VALUE)
- priv->next_pts = mpframe->pts + mpframe->samples / (double)mpframe->rate;
+ priv->next_pts = mp_aframe_end_pts(mpframe);
#if LIBAVCODEC_VERSION_MICRO >= 100
AVFrameSideData *sd =
@@ -254,14 +246,14 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
priv->preroll_done = true;
}
- uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
+ uint32_t skip = MPMIN(priv->skip_samples, mp_aframe_get_size(mpframe));
if (skip) {
- mp_audio_skip_samples(mpframe, skip);
+ mp_aframe_skip_samples(mpframe, skip);
priv->skip_samples -= skip;
}
- uint32_t trim = MPMIN(priv->trim_samples, mpframe->samples);
+ uint32_t trim = MPMIN(priv->trim_samples, mp_aframe_get_size(mpframe));
if (trim) {
- mpframe->samples -= trim;
+ mp_aframe_set_size(mpframe, mp_aframe_get_size(mpframe) - trim);
priv->trim_samples -= trim;
}
@@ -269,7 +261,6 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
av_frame_unref(priv->avframe);
- MP_DBG(da, "Decoded %d samples\n", mpframe->samples);
return true;
}
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 4b3e8149ec..0ca20e5485 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -40,8 +40,9 @@ struct spdifContext {
uint8_t out_buffer[OUTBUF_SIZE];
bool need_close;
bool use_dts_hd;
- struct mp_audio fmt;
- struct mp_audio_pool *pool;
+ struct mp_aframe *fmt;
+ int sstride;
+ struct mp_aframe_pool *pool;
bool got_eof;
struct demux_packet *queued_packet;
};
@@ -84,7 +85,7 @@ static int init(struct dec_audio *da, const char *decoder)
da->priv = spdif_ctx;
spdif_ctx->log = da->log;
spdif_ctx->use_dts_hd = da->opts->dtshd;
- spdif_ctx->pool = mp_audio_pool_create(spdif_ctx);
+ spdif_ctx->pool = mp_aframe_pool_create(spdif_ctx);
if (strcmp(decoder, "spdif_dts_hd") == 0)
spdif_ctx->use_dts_hd = true;
@@ -198,6 +199,9 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
AVDictionary *format_opts = NULL;
+ spdif_ctx->fmt = mp_aframe_create();
+ talloc_steal(spdif_ctx, spdif_ctx->fmt);
+
int num_channels = 0;
int sample_format = 0;
int samplerate = 0;
@@ -246,9 +250,14 @@ static int init_filter(struct dec_audio *da, AVPacket *pkt)
default:
abort();
}
- mp_audio_set_num_channels(&spdif_ctx->fmt, num_channels);
- mp_audio_set_format(&spdif_ctx->fmt, sample_format);
- spdif_ctx->fmt.rate = samplerate;
+
+ struct mp_chmap chmap;
+ mp_chmap_from_channels(&chmap, num_channels);
+ mp_aframe_set_chmap(spdif_ctx->fmt, &chmap);
+ mp_aframe_set_format(spdif_ctx->fmt, sample_format);
+ mp_aframe_set_rate(spdif_ctx->fmt, samplerate);
+
+ spdif_ctx->sstride = mp_aframe_get_sstride(spdif_ctx->fmt);
if (avformat_write_header(lavf_ctx, &format_opts) < 0) {
MP_FATAL(da, "libavformat spdif initialization failed.\n");
@@ -279,7 +288,7 @@ static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
return true;
}
-static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
+static bool receive_frame(struct dec_audio *da, struct mp_aframe **out)
{
struct spdifContext *spdif_ctx = da->priv;
@@ -308,13 +317,21 @@ static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
goto done;
}
- int samples = spdif_ctx->out_buffer_len / spdif_ctx->fmt.sstride;
- *out = mp_audio_pool_get(spdif_ctx->pool, &spdif_ctx->fmt, samples);
- if (!*out)
+ *out = mp_aframe_new_ref(spdif_ctx->fmt);
+ int samples = spdif_ctx->out_buffer_len / spdif_ctx->sstride;
+ if (mp_aframe_pool_allocate(spdif_ctx->pool, *out, samples) < 0) {
+ TA_FREEP(out);
goto done;
+ }
+
+ uint8_t **data = mp_aframe_get_data_rw(*out);
+ if (!data) {
+ TA_FREEP(out);
+ goto done;
+ }
- memcpy((*out)->planes[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
- (*out)->pts = pts;
+ memcpy(data[0], spdif_ctx->out_buffer, spdif_ctx->out_buffer_len);
+ mp_aframe_set_pts(*out, pts);
done:
talloc_free(spdif_ctx->queued_packet);
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index 1351cb8ecd..401e26fb7b 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -38,8 +38,6 @@
#include "dec_audio.h"
#include "ad.h"
#include "audio/format.h"
-#include "audio/audio.h"
-#include "audio/audio_buffer.h"
#include "audio/filter/af.h"
@@ -179,25 +177,24 @@ static void fix_audio_pts(struct dec_audio *da)
if (!da->current_frame)
return;
- if (da->current_frame->pts != MP_NOPTS_VALUE) {
- double newpts = da->current_frame->pts;
-
+ double frame_pts = mp_aframe_get_pts(da->current_frame);
+ if (frame_pts != MP_NOPTS_VALUE) {
if (da->pts != MP_NOPTS_VALUE)
- MP_STATS(da, "value %f audio-pts-err", da->pts - newpts);
+ MP_STATS(da, "value %f audio-pts-err", da->pts - frame_pts);
// Keep the interpolated timestamp if it doesn't deviate more
// than 1 ms from the real one. (MKV rounded timestamps.)
- if (da->pts == MP_NOPTS_VALUE || fabs(da->pts - newpts) > 0.001)
- da->pts = newpts;
+ if (da->pts == MP_NOPTS_VALUE || fabs(da->pts - frame_pts) > 0.001)
+ da->pts = frame_pts;
}
if (da->pts == MP_NOPTS_VALUE && da->header->missing_timestamps)
da->pts = 0;
- da->current_frame->pts = da->pts;
+ mp_aframe_set_pts(da->current_frame, da->pts);
if (da->pts != MP_NOPTS_VALUE)
- da->pts += da->current_frame->samples / (double)da->current_frame->rate;
+ da->pts += mp_aframe_duration(da->current_frame);
}
void audio_work(struct dec_audio *da)
@@ -228,11 +225,6 @@ void audio_work(struct dec_audio *da)
bool progress = da->ad_driver->receive_frame(da, &da->current_frame);
- if (da->current_frame && !mp_audio_config_valid(da->current_frame)) {
- talloc_free(da->current_frame);
- da->current_frame = NULL;
- }
-
da->current_state = da->current_frame ? DATA_OK : DATA_AGAIN;
if (!progress)
da->current_state = DATA_EOF;
@@ -242,10 +234,11 @@ void audio_work(struct dec_audio *da)
bool segment_end = da->current_state == DATA_EOF;
if (da->current_frame) {
- mp_audio_clip_timestamps(da->current_frame, da->start, da->end);
- if (da->current_frame->pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
- segment_end = da->current_frame->pts >= da->end;
- if (da->current_frame->samples == 0) {
+ mp_aframe_clip_timestamps(da->current_frame, da->start, da->end);
+ double frame_pts = mp_aframe_get_pts(da->current_frame);
+ if (frame_pts != MP_NOPTS_VALUE && da->start != MP_NOPTS_VALUE)
+ segment_end = frame_pts >= da->end;
+ if (mp_aframe_get_size(da->current_frame) == 0) {
talloc_free(da->current_frame);
da->current_frame = NULL;
}
@@ -280,7 +273,7 @@ void audio_work(struct dec_audio *da)
// DATA_WAIT: waiting for demuxer; will receive a wakeup signal
// DATA_EOF: end of file, no more frames to be expected
// DATA_AGAIN: dropped frame or something similar
-int audio_get_frame(struct dec_audio *da, struct mp_audio **out_frame)
+int audio_get_frame(struct dec_audio *da, struct mp_aframe **out_frame)
{
*out_frame = NULL;
if (da->current_frame) {
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
index 886b617b58..ea504328df 100644
--- a/audio/decode/dec_audio.h
+++ b/audio/decode/dec_audio.h
@@ -19,11 +19,10 @@
#define MPLAYER_DEC_AUDIO_H
#include "audio/chmap.h"
-#include "audio/audio.h"
+#include "audio/aframe.h"
#include "demux/demux.h"
#include "demux/stheader.h"
-struct mp_audio_buffer;
struct mp_decoder_list;
struct dec_audio {
@@ -48,7 +47,7 @@ struct dec_audio {
double start, end;
struct demux_packet *packet;
struct demux_packet *new_segment;
- struct mp_audio *current_frame;
+ struct mp_aframe *current_frame;
int current_state;
};
@@ -57,7 +56,7 @@ int audio_init_best_codec(struct dec_audio *d_audio);
void audio_uninit(struct dec_audio *d_audio);
void audio_work(struct dec_audio *d_audio);
-int audio_get_frame(struct dec_audio *d_audio, struct mp_audio **out_frame);
+int audio_get_frame(struct dec_audio *d_audio, struct mp_aframe **out_frame);
void audio_reset_decoding(struct dec_audio *d_audio);
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-27 10:45:09 +0000