summaryrefslogtreecommitdiff
path: root/libavcodec/aacdec.c
diff options
context:
space:
mode:
Diffstat (limited to 'libavcodec/aacdec.c')
-rw-r--r--libavcodec/aacdec.c106
1 files changed, 61 insertions, 45 deletions
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 5a2b230d24..a046d991e6 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -471,15 +471,17 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
* @param ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
- * @param data pointer to AVCodecContext extradata
- * @param data_size size of AVCCodecContext extradata
+ * @param data pointer to buffer holding an audio specific config
+ * @param bit_size size of audio specific config or data in bits
+ * @param sync_extension look for an appended sync extension
*
* @return Returns error status or number of consumed bits. <0 - error
*/
static int decode_audio_specific_config(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
- const uint8_t *data, int data_size, int asclen)
+ const uint8_t *data, int bit_size,
+ int sync_extension)
{
GetBitContext gb;
int i;
@@ -489,9 +491,9 @@ static int decode_audio_specific_config(AACContext *ac,
av_dlog(avctx, "%02x ", avctx->extradata[i]);
av_dlog(avctx, "\n");
- init_get_bits(&gb, data, data_size * 8);
+ init_get_bits(&gb, data, bit_size);
- if ((i = avpriv_mpeg4audio_get_config(m4ac, data, asclen/8)) < 0)
+ if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
return -1;
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
@@ -591,7 +593,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
if (avctx->extradata_size > 0) {
if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
avctx->extradata,
- avctx->extradata_size, 8*avctx->extradata_size) < 0)
+ avctx->extradata_size*8, 1) < 0)
return -1;
} else {
int sr, i;
@@ -665,6 +667,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
cbrt_tableinit();
+ avcodec_get_frame_defaults(&ac->frame);
+ avctx->coded_frame = &ac->frame;
+
return 0;
}
@@ -2132,12 +2137,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
- int *data_size, GetBitContext *gb)
+ int *got_frame_ptr, GetBitContext *gb)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
- int err, elem_id, data_size_tmp;
+ int err, elem_id;
int samples = 0, multiplier, audio_found = 0;
if (show_bits(gb, 12) == 0xfff) {
@@ -2250,24 +2255,26 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
avctx->frame_size = samples;
}
- data_size_tmp = samples * avctx->channels *
- av_get_bytes_per_sample(avctx->sample_fmt);
- if (*data_size < data_size_tmp) {
- av_log(avctx, AV_LOG_ERROR,
- "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
- *data_size, data_size_tmp);
- return -1;
- }
- *data_size = data_size_tmp;
-
if (samples) {
+ /* get output buffer */
+ ac->frame.nb_samples = samples;
+ if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return err;
+ }
+
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
- ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
+ ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
+ (const float **)ac->output_data,
samples, avctx->channels);
else
- ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
+ ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
+ (const float **)ac->output_data,
samples, avctx->channels);
+
+ *(AVFrame *)data = ac->frame;
}
+ *got_frame_ptr = !!samples;
if (ac->output_configured && audio_found)
ac->output_configured = OC_LOCKED;
@@ -2276,7 +2283,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
}
static int aac_decode_frame(AVCodecContext *avctx, void *data,
- int *data_size, AVPacket *avpkt)
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@@ -2287,7 +2294,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
init_get_bits(&gb, buf, buf_size * 8);
- if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@@ -2340,30 +2347,40 @@ static inline uint32_t latm_get_value(GetBitContext *b)
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
GetBitContext *gb, int asclen)
{
- AVCodecContext *avctx = latmctx->aac_ctx.avctx;
- AACContext *ac= &latmctx->aac_ctx;
- MPEG4AudioConfig m4ac=ac->m4ac;
- int config_start_bit = get_bits_count(gb);
- int bits_consumed, esize;
+ AACContext *ac = &latmctx->aac_ctx;
+ AVCodecContext *avctx = ac->avctx;
+ MPEG4AudioConfig m4ac = {0};
+ int config_start_bit = get_bits_count(gb);
+ int sync_extension = 0;
+ int bits_consumed, esize;
+
+ if (asclen) {
+ sync_extension = 1;
+ asclen = FFMIN(asclen, get_bits_left(gb));
+ } else
+ asclen = get_bits_left(gb);
if (config_start_bit % 8) {
av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
"config not byte aligned.\n", 1);
return AVERROR_INVALIDDATA;
- } else {
- bits_consumed =
- decode_audio_specific_config(ac, avctx, &m4ac,
+ }
+ bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
gb->buffer + (config_start_bit / 8),
- get_bits_left(gb) / 8, asclen);
+ asclen, sync_extension);
- if (bits_consumed < 0)
- return AVERROR_INVALIDDATA;
- if(ac->m4ac.sample_rate != m4ac.sample_rate || m4ac.chan_config != ac->m4ac.chan_config)
- ac->m4ac= m4ac;
+ if (bits_consumed < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (ac->m4ac.sample_rate != m4ac.sample_rate ||
+ ac->m4ac.chan_config != m4ac.chan_config) {
+
+ av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+ latmctx->initialized = 0;
esize = (bits_consumed+7) / 8;
- if (avctx->extradata_size <= esize) {
+ if (avctx->extradata_size < esize) {
av_free(avctx->extradata);
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
@@ -2373,9 +2390,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
avctx->extradata_size = esize;
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
-
- skip_bits_long(gb, bits_consumed);
}
+ skip_bits_long(gb, bits_consumed);
return bits_consumed;
}
@@ -2512,8 +2528,8 @@ static int read_audio_mux_element(struct LATMContext *latmctx,
}
-static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
- AVPacket *avpkt)
+static int latm_decode_frame(AVCodecContext *avctx, void *out,
+ int *got_frame_ptr, AVPacket *avpkt)
{
struct LATMContext *latmctx = avctx->priv_data;
int muxlength, err;
@@ -2535,12 +2551,12 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
if (!latmctx->initialized) {
if (!avctx->extradata) {
- *out_size = 0;
+ *got_frame_ptr = 0;
return avpkt->size;
} else {
if ((err = decode_audio_specific_config(
&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.m4ac,
- avctx->extradata, avctx->extradata_size, 8*avctx->extradata_size)) < 0)
+ avctx->extradata, avctx->extradata_size*8, 1)) < 0)
return err;
latmctx->initialized = 1;
}
@@ -2553,7 +2569,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
return AVERROR_INVALIDDATA;
}
- if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0)
return err;
return muxlength;
@@ -2583,7 +2599,7 @@ AVCodec ff_aac_decoder = {
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
- .capabilities = CODEC_CAP_CHANNEL_CONF,
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};
@@ -2604,7 +2620,7 @@ AVCodec ff_aac_latm_decoder = {
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
- .capabilities = CODEC_CAP_CHANNEL_CONF,
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
.flush = flush,
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-30 22:39:21 +0000