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-rw-r--r--audio/decode/ad.h2
-rw-r--r--audio/decode/ad_lavc.c69
-rw-r--r--audio/decode/ad_mpg123.c210
-rw-r--r--audio/decode/ad_spdif.c27
-rw-r--r--audio/decode/dec_audio.c53
-rw-r--r--audio/decode/dec_audio.h5
6 files changed, 127 insertions, 239 deletions
diff --git a/audio/decode/ad.h b/audio/decode/ad.h
index 2ac9fb21f6..e09ded2efc 100644
--- a/audio/decode/ad.h
+++ b/audio/decode/ad.h
@@ -36,7 +36,7 @@ struct ad_functions {
int (*init)(struct dec_audio *da, const char *decoder);
void (*uninit)(struct dec_audio *da);
int (*control)(struct dec_audio *da, int cmd, void *arg);
- int (*decode_audio)(struct dec_audio *da, struct mp_audio *buffer, int maxlen);
+ int (*decode_packet)(struct dec_audio *da);
};
enum ad_ctrl {
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 4d892da32b..cb8cfa8c82 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -49,7 +49,6 @@ struct priv {
};
static void uninit(struct dec_audio *da);
-static int decode_new_packet(struct dec_audio *da);
#define OPT_BASE_STRUCT struct ad_lavc_params
struct ad_lavc_params {
@@ -143,11 +142,10 @@ static const char *find_pcm_decoder(const struct pcm_map *map, int format,
return NULL;
}
-static int setup_format(struct dec_audio *da)
+static void set_data_from_avframe(struct dec_audio *da)
{
struct priv *priv = da->priv;
AVCodecContext *lavc_context = priv->avctx;
- struct sh_audio *sh_audio = da->header->audio;
// Note: invalid parameters are rejected by dec_audio.c
@@ -164,12 +162,15 @@ static int setup_format(struct dec_audio *da)
if (lavc_chmap.num != lavc_context->channels)
mp_chmap_from_channels(&lavc_chmap, lavc_context->channels);
if (priv->force_channel_map) {
+ struct sh_audio *sh_audio = da->header->audio;
if (lavc_chmap.num == sh_audio->channels.num)
lavc_chmap = sh_audio->channels;
}
mp_audio_set_channels(&da->decoded, &lavc_chmap);
- return 0;
+ da->decoded.samples = priv->avframe->nb_samples;
+ for (int n = 0; n < da->decoded.num_planes; n++)
+ da->decoded.planes[n] = priv->avframe->data[n];
}
static void set_from_wf(AVCodecContext *avctx, MP_WAVEFORMATEX *wf)
@@ -261,22 +262,6 @@ static int init(struct dec_audio *da, const char *decoder)
uninit(da);
return 0;
}
- MP_VERBOSE(da, "INFO: libavcodec \"%s\" init OK!\n",
- lavc_codec->name);
-
- // Decode at least 1 sample: (to get header filled)
- for (int tries = 1; ; tries++) {
- int x = decode_new_packet(da);
- if (x >= 0 && ctx->frame.samples > 0) {
- MP_VERBOSE(da, "Initial decode succeeded after %d packets.\n", tries);
- break;
- }
- if (tries >= 50) {
- MP_ERR(da, "initial decode failed\n");
- uninit(da);
- return 0;
- }
- }
if (lavc_context->bit_rate != 0)
da->bitrate = lavc_context->bit_rate;
@@ -308,7 +293,7 @@ static int control(struct dec_audio *da, int cmd, void *arg)
switch (cmd) {
case ADCTRL_RESET:
avcodec_flush_buffers(ctx->avctx);
- ctx->frame.samples = 0;
+ mp_audio_set_null_data(&da->decoded);
talloc_free(ctx->packet);
ctx->packet = NULL;
return CONTROL_TRUE;
@@ -316,12 +301,12 @@ static int control(struct dec_audio *da, int cmd, void *arg)
return CONTROL_UNKNOWN;
}
-static int decode_new_packet(struct dec_audio *da)
+static int decode_packet(struct dec_audio *da)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
- priv->frame.samples = 0;
+ mp_audio_set_null_data(&da->decoded);
struct demux_packet *mpkt = priv->packet;
if (!mpkt)
@@ -361,19 +346,13 @@ static int decode_new_packet(struct dec_audio *da)
return 0;
}
if (ret < 0) {
- MP_VERBOSE(da, "lavc_audio: error\n");
- return -1;
+ MP_ERR(da, "Error decoding audio.\n");
+ return AD_ERR;
}
if (!got_frame)
- return mpkt ? 0 : -1; // -1: eof
-
- if (setup_format(da) < 0)
- return -1;
+ return mpkt ? AD_OK : AD_EOF;
- priv->frame.samples = priv->avframe->nb_samples;
- mp_audio_copy_config(&priv->frame, &da->decoded);
- for (int n = 0; n < priv->frame.num_planes; n++)
- priv->frame.planes[n] = priv->avframe->data[n];
+ set_data_from_avframe(da);
double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
if (out_pts != MP_NOPTS_VALUE) {
@@ -381,27 +360,7 @@ static int decode_new_packet(struct dec_audio *da)
da->pts_offset = 0;
}
- MP_DBG(da, "Decoded %d -> %d samples\n", in_len,
- priv->frame.samples);
- return 0;
-}
-
-static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxlen)
-{
- struct priv *priv = da->priv;
-
- if (!priv->frame.samples) {
- if (decode_new_packet(da) < 0)
- return AD_ERR;
- }
-
- if (!mp_audio_config_equals(buffer, &priv->frame))
- return 0;
-
- buffer->samples = MPMIN(priv->frame.samples, maxlen);
- mp_audio_copy(buffer, 0, &priv->frame, 0, buffer->samples);
- mp_audio_skip_samples(&priv->frame, buffer->samples);
- da->pts_offset += buffer->samples;
+ MP_DBG(da, "Decoded %d -> %d samples\n", in_len, da->decoded.samples);
return 0;
}
@@ -418,5 +377,5 @@ const struct ad_functions ad_lavc = {
.init = init,
.uninit = uninit,
.control = control,
- .decode_audio = decode_audio,
+ .decode_packet = decode_packet,
};
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index 055285cccd..f96a5a8036 100644
--- a/audio/decode/ad_mpg123.c
+++ b/audio/decode/ad_mpg123.c
@@ -35,9 +35,7 @@
struct ad_mpg123_context {
mpg123_handle *handle;
- bool new_format;
int sample_size;
- bool need_data;
/* Running mean for bit rate, stream length estimation. */
float mean_rate;
unsigned int mean_count;
@@ -58,7 +56,7 @@ static void uninit(struct dec_audio *da)
/* This initializes libmpg123 and prepares the handle, including funky
* parameters. */
-static int preinit(struct dec_audio *da)
+static int init(struct dec_audio *da, const char *decoder)
{
int err;
struct ad_mpg123_context *con;
@@ -111,15 +109,18 @@ static int preinit(struct dec_audio *da)
* We need at least 1152 samples. dec_audio.c normally guarantees this. */
mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.);
+ err = mpg123_open_feed(con->handle);
+ if (err != MPG123_OK)
+ goto bad_end;
+
return 1;
bad_end:
- if (!con->handle)
- MP_ERR(da, "mpg123 preinit error: %s\n",
- mpg123_plain_strerror(err));
- else
- MP_ERR(da, "mpg123 preinit error: %s\n",
- mpg123_strerror(con->handle));
+ if (!con->handle) {
+ MP_ERR(da, "mpg123 preinit error: %s\n", mpg123_plain_strerror(err));
+ } else {
+ MP_ERR(da, "mpg123 preinit error: %s\n", mpg123_strerror(con->handle));
+ }
uninit(da);
return 0;
@@ -140,111 +141,6 @@ static int mpg123_format_to_af(int mpg123_encoding)
return 0;
}
-/* libmpg123 has a new format ready; query and store, return return value
- of mpg123_getformat() */
-static int set_format(struct dec_audio *da)
-{
- struct ad_mpg123_context *con = da->priv;
- int ret;
- long rate;
- int channels;
- int encoding;
- ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
- if (ret == MPG123_OK) {
- mp_audio_set_num_channels(&da->decoded, channels);
- da->decoded.rate = rate;
- int af = mpg123_format_to_af(encoding);
- if (!af) {
- /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
- MP_ERR(da, "Bad encoding from mpg123: %i.\n", encoding);
- return MPG123_ERR;
- }
- mp_audio_set_format(&da->decoded, af);
- con->sample_size = channels * af_fmt2bps(af);
- con->new_format = 0;
- }
- return ret;
-}
-
-static int feed_new_packet(struct dec_audio *da)
-{
- struct ad_mpg123_context *con = da->priv;
- int ret;
-
- struct demux_packet *pkt = demux_read_packet(da->header);
- if (!pkt)
- return -1; /* EOF. */
-
- /* Next bytes from that presentation time. */
- if (pkt->pts != MP_NOPTS_VALUE) {
- da->pts = pkt->pts;
- da->pts_offset = 0;
- }
-
- /* Have to use mpg123_feed() to avoid decoding here. */
- ret = mpg123_feed(con->handle, pkt->buffer, pkt->len);
- talloc_free(pkt);
-
- if (ret == MPG123_ERR)
- return -1;
-
- if (ret == MPG123_NEW_FORMAT)
- con->new_format = 1;
-
- return 0;
-}
-
-/* Now we really start accessing some data and determining file format.
- * Format now is allowed to change on-the-fly. Here is the only point
- * that has MPlayer react to errors. We have to pray that exceptional
- * erros in other places simply cannot occur. */
-static int init(struct dec_audio *da, const char *decoder)
-{
- if (!preinit(da))
- return 0;
-
- struct ad_mpg123_context *con = da->priv;
- int ret;
-
- ret = mpg123_open_feed(con->handle);
- if (ret != MPG123_OK)
- goto fail;
-
- for (int n = 0; ; n++) {
- if (feed_new_packet(da) < 0) {
- ret = MPG123_NEED_MORE;
- goto fail;
- }
- size_t got_now = 0;
- ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
- if (ret == MPG123_OK || ret == MPG123_NEW_FORMAT) {
- ret = set_format(da);
- if (ret == MPG123_OK)
- break;
- }
- if (ret != MPG123_NEED_MORE)
- goto fail;
- // max. 16 retries (randomly chosen number)
- if (n > 16) {
- ret = MPG123_NEED_MORE;
- goto fail;
- }
- }
-
- return 1;
-
-fail:
- if (ret == MPG123_NEED_MORE) {
- MP_ERR(da, "Could not find mp3 stream.\n");
- } else {
- MP_ERR(da, "mpg123 init error: %s\n",
- mpg123_strerror(con->handle));
- }
-
- uninit(da);
- return 0;
-}
-
/* Compute bitrate from frame size. */
static int compute_bitrate(struct mpg123_frameinfo *i)
{
@@ -290,50 +186,79 @@ static void update_info(struct dec_audio *da)
}
}
-static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxlen)
+/* libmpg123 has a new format ready; query and store, return return value
+ of mpg123_getformat() */
+static int set_format(struct dec_audio *da)
{
struct ad_mpg123_context *con = da->priv;
- void *buf = buffer->planes[0];
int ret;
-
- if (con->new_format) {
- ret = set_format(da);
- if (ret == MPG123_OK) {
- return 0; // let caller handle format change
- } else if (ret == MPG123_NEED_MORE) {
- con->need_data = true;
- } else {
- goto mpg123_fail;
+ long rate;
+ int channels;
+ int encoding;
+ ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
+ if (ret == MPG123_OK) {
+ mp_audio_set_num_channels(&da->decoded, channels);
+ da->decoded.rate = rate;
+ int af = mpg123_format_to_af(encoding);
+ if (!af) {
+ /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
+ MP_ERR(da, "Bad encoding from mpg123: %i.\n", encoding);
+ return MPG123_ERR;
}
+ mp_audio_set_format(&da->decoded, af);
+ con->sample_size = channels * af_fmt2bps(af);
}
+ return ret;
+}
- if (con->need_data) {
- if (feed_new_packet(da) < 0)
- return AD_ERR;
+static int decode_packet(struct dec_audio *da)
+{
+ struct ad_mpg123_context *con = da->priv;
+ int ret;
+
+ mp_audio_set_null_data(&da->decoded);
+
+ struct demux_packet *pkt = demux_read_packet(da->header);
+ if (!pkt)
+ return AD_EOF;
+
+ /* Next bytes from that presentation time. */
+ if (pkt->pts != MP_NOPTS_VALUE) {
+ da->pts = pkt->pts;
+ da->pts_offset = 0;
}
- if (!mp_audio_config_equals(&da->decoded, buffer))
- return 0;
+ /* Have to use mpg123_feed() to avoid decoding here. */
+ ret = mpg123_feed(con->handle, pkt->buffer, pkt->len);
+ talloc_free(pkt);
- size_t got_now = 0;
- ret = mpg123_replace_buffer(con->handle, buf, maxlen * con->sample_size);
if (ret != MPG123_OK)
goto mpg123_fail;
- ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
+ unsigned char *audio = NULL;
+ size_t bytes = 0;
+ ret = mpg123_decode_frame(con->handle, NULL, &audio, &bytes);
- int got_samples = got_now / con->sample_size;
- buffer->samples += got_samples;
- da->pts_offset += got_samples;
+ if (ret == MPG123_NEED_MORE)
+ return 0;
+
+ if (ret != MPG123_OK && ret != MPG123_DONE && ret != MPG123_NEW_FORMAT)
+ goto mpg123_fail;
- if (ret == MPG123_NEW_FORMAT) {
- con->new_format = true;
- } else if (ret == MPG123_NEED_MORE) {
- con->need_data = true;
- } else if (ret != MPG123_OK && ret != MPG123_DONE) {
+ ret = set_format(da);
+ if (ret != MPG123_OK)
goto mpg123_fail;
+
+ if (con->sample_size < 1) {
+ MP_ERR(da, "no sample size\n");
+ return AD_ERR;
}
+ int got_samples = bytes / con->sample_size;
+ da->decoded.planes[0] = audio;
+ da->decoded.samples = got_samples;
+ da->pts_offset += got_samples;
+
update_info(da);
return 0;
@@ -348,6 +273,7 @@ static int control(struct dec_audio *da, int cmd, void *arg)
switch (cmd) {
case ADCTRL_RESET:
+ mp_audio_set_null_data(&da->decoded);
mpg123_close(con->handle);
if (mpg123_open_feed(con->handle) != MPG123_OK) {
@@ -372,5 +298,5 @@ const struct ad_functions ad_mpg123 = {
.init = init,
.uninit = uninit,
.control = control,
- .decode_audio = decode_audio,
+ .decode_packet = decode_packet,
};
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 1042fdeb48..d655f91574 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -38,8 +38,7 @@ struct spdifContext {
AVFormatContext *lavf_ctx;
int iec61937_packet_size;
int out_buffer_len;
- int out_buffer_size;
- uint8_t *out_buffer;
+ uint8_t out_buffer[OUTBUF_SIZE];
bool need_close;
};
@@ -47,7 +46,7 @@ static int write_packet(void *p, uint8_t *buf, int buf_size)
{
struct spdifContext *ctx = p;
- int buffer_left = ctx->out_buffer_size - ctx->out_buffer_len;
+ int buffer_left = OUTBUF_SIZE - ctx->out_buffer_len;
if (buf_size > buffer_left) {
MP_ERR(ctx, "spdif packet too large.\n");
buf_size = buffer_left;
@@ -183,24 +182,18 @@ fail:
return 0;
}
-static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxlen)
+static int decode_packet(struct dec_audio *da)
{
struct spdifContext *spdif_ctx = da->priv;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
- int sstride = 2 * da->decoded.channels.num;
- assert(sstride == buffer->sstride);
-
- if (maxlen * sstride < spdif_ctx->iec61937_packet_size)
- return 0;
+ mp_audio_set_null_data(&da->decoded);
spdif_ctx->out_buffer_len = 0;
- spdif_ctx->out_buffer_size = maxlen * sstride;
- spdif_ctx->out_buffer = buffer->planes[0];
struct demux_packet *mpkt = demux_read_packet(da->header);
if (!mpkt)
- return AD_ERR;
+ return AD_EOF;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
@@ -211,13 +204,15 @@ static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxle
da->pts_offset = 0;
}
int ret = av_write_frame(lavf_ctx, &pkt);
- avio_flush(lavf_ctx->pb);
- buffer->samples = spdif_ctx->out_buffer_len / sstride;
- da->pts_offset += buffer->samples;
talloc_free(mpkt);
+ avio_flush(lavf_ctx->pb);
if (ret < 0)
return AD_ERR;
+ da->decoded.planes[0] = spdif_ctx->out_buffer;
+ da->decoded.samples = spdif_ctx->out_buffer_len / da->decoded.sstride;
+ da->pts_offset += da->decoded.samples;
+
return 0;
}
@@ -253,5 +248,5 @@ const struct ad_functions ad_spdif = {
.init = init,
.uninit = uninit,
.control = control,
- .decode_audio = decode_audio,
+ .decode_packet = decode_packet,
};
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index 907b154bf8..c2857353fa 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -56,13 +56,6 @@ static const struct ad_functions * const ad_drivers[] = {
NULL
};
-// ad_mpg123 needs to be able to decode 1152 samples at once
-// ad_spdif needs up to 8192
-#define DECODE_MAX_UNIT MPMAX(8192, 1152)
-
-// At least 8192 samples, plus hack for ad_mpg123 and ad_spdif
-#define DECODE_BUFFER_SAMPLES (8192 + DECODE_MAX_UNIT)
-
// Drop audio buffer and reinit it (after format change)
// Returns whether the format was valid at all.
static bool reinit_audio_buffer(struct dec_audio *da)
@@ -73,7 +66,6 @@ static bool reinit_audio_buffer(struct dec_audio *da)
return false;
}
mp_audio_buffer_reinit(da->decode_buffer, &da->decoded);
- mp_audio_buffer_preallocate_min(da->decode_buffer, DECODE_BUFFER_SAMPLES);
return true;
}
@@ -97,6 +89,21 @@ static int init_audio_codec(struct dec_audio *d_audio, const char *decoder)
return 0;
}
+ // Decode enough until we know the audio format.
+ for (int tries = 1; ; tries++) {
+ if (mp_audio_config_valid(&d_audio->decoded)) {
+ MP_VERBOSE(d_audio, "Initial decode succeeded after %d packets.\n",
+ tries);
+ break;
+ }
+ if (tries >= 50) {
+ MP_ERR(d_audio, "initial decode failed\n");
+ uninit_decoder(d_audio);
+ return 0;
+ }
+ d_audio->ad_driver->decode_packet(d_audio);
+ }
+
d_audio->decode_buffer = mp_audio_buffer_create(NULL);
if (!reinit_audio_buffer(d_audio)) {
uninit_decoder(d_audio);
@@ -241,26 +248,28 @@ static int filter_n_bytes(struct dec_audio *da, struct mp_audio_buffer *outbuf,
mp_audio_buffer_get_format(da->decode_buffer, &config);
while (mp_audio_buffer_samples(da->decode_buffer) < len) {
- int maxlen = mp_audio_buffer_get_write_available(da->decode_buffer);
- if (maxlen < DECODE_MAX_UNIT)
- break;
- struct mp_audio buffer;
- mp_audio_buffer_get_write_buffer(da->decode_buffer, maxlen, &buffer);
- buffer.samples = 0;
- error = da->ad_driver->decode_audio(da, &buffer, maxlen);
- if (error < 0)
- break;
- // Commit the data just read as valid data
- mp_audio_buffer_finish_write(da->decode_buffer, buffer.samples);
// Format change
if (!mp_audio_config_equals(&da->decoded, &config)) {
// If there are still samples left in the buffer, let them drain
// first, and don't signal a format change to the caller yet.
- if (mp_audio_buffer_samples(da->decode_buffer) > 0)
- break;
- error = AD_NEW_FMT;
+ if (mp_audio_buffer_samples(da->decode_buffer) == 0)
+ error = AD_NEW_FMT;
break;
}
+ if (da->decoded.samples > 0) {
+ int copy = MPMIN(da->decoded.samples, len);
+ struct mp_audio append = da->decoded;
+ append.samples = copy;
+ mp_audio_buffer_append(da->decode_buffer, &append);
+ mp_audio_skip_samples(&da->decoded, copy);
+ continue;
+ }
+ error = da->ad_driver->decode_packet(da);
+ if (error < 0)
+ break;
+ // No progress means the decoder is buffering input data.
+ if (!da->decoded.samples)
+ break;
}
// Filter
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
index 25afda1390..c1b5eafb49 100644
--- a/audio/decode/dec_audio.h
+++ b/audio/decode/dec_audio.h
@@ -38,9 +38,7 @@ struct dec_audio {
char *decoder_desc;
struct replaygain_data *replaygain_data;
// set by decoder
- struct mp_audio decoded; // format of decoded audio (no data, temporarily
- // different from decode_buffer during format
- // changes)
+ struct mp_audio decoded; // decoded audio set by last decode_packet() call
int bitrate; // input bitrate, can change with VBR sources
// last known pts value in output from decoder
double pts;
@@ -53,6 +51,7 @@ struct dec_audio {
enum {
AD_OK = 0,
AD_ERR = -1,
+ AD_EOF = -1, // same as AD_ERR for now
AD_NEW_FMT = -2,
AD_ASYNC_PLAY_DONE = -3,
};
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-01 02:51:34 +0000