summaryrefslogtreecommitdiffstats
path: root/audio/filter
diff options
context:
space:
mode:
authorwm4 <wm4@nowhere>2015-01-14 22:16:08 +0100
committerwm4 <wm4@nowhere>2015-01-14 22:16:30 +0100
commite865d255d0a61abe5be8fad27b2f5e59381cb53a (patch)
tree12805804be143372681bb8edc2ca4750d17f2e68 /audio/filter
parent5d972491bbfc38d9fd9031de2841eed7feeea422 (diff)
downloadmpv-e865d255d0a61abe5be8fad27b2f5e59381cb53a.tar.bz2
mpv-e865d255d0a61abe5be8fad27b2f5e59381cb53a.tar.xz
af_lavcac3enc: use refcounted frames
Diffstat (limited to 'audio/filter')
-rw-r--r--audio/filter/af_lavcac3enc.c184
1 files changed, 95 insertions, 89 deletions
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index d4f559ad07..3ced241c62 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -53,7 +53,8 @@ typedef struct af_ac3enc_s {
struct AVCodecContext *lavc_actx;
AVPacket pkt;
int bit_rate;
- struct mp_audio_buffer *pending;
+ struct mp_audio *input; // frame passed to libavcodec
+ struct mp_audio *pending; // unconsumed input data
int in_samples; // samples of input per AC3 frame
int out_samples; // upper bound on encoded output per AC3 frame
int in_sampleformat;
@@ -101,7 +102,12 @@ static int control(struct af_instance *af, int cmd, void *arg)
s->out_samples = AC3_MAX_CODED_FRAME_SIZE / af->data->sstride;
}
- mp_audio_buffer_reinit(s->pending, in);
+ mp_audio_copy_config(s->input, in);
+ mp_audio_realloc(s->input, s->in_samples);
+ s->input->samples = 0;
+
+ talloc_free(s->pending);
+ s->pending = NULL;
MP_DBG(af, "af_lavcac3enc reinit: %d, %d, %d.\n",
in->nch, in->rate, s->in_samples);
@@ -147,111 +153,110 @@ static void uninit(struct af_instance* af)
avcodec_close(s->lavc_actx);
av_free(s->lavc_actx);
}
+ talloc_free(s->pending);
}
}
+static int filter_frame(struct af_instance *af, struct mp_audio *audio)
+{
+ af_ac3enc_t *s = af->priv;
+
+ // filter_output must have been called until no output was produced.
+ if (s->pending && s->pending->samples)
+ MP_ERR(af, "broken data flow\n");
+
+ talloc_free(s->pending);
+ s->pending = audio;
+ return 0;
+}
+
static void swap_16(uint16_t *ptr, size_t size)
{
for (size_t n = 0; n < size; n++)
ptr[n] = av_bswap16(ptr[n]);
}
-// Filter data through filter
-static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
+// Copy data from input frame to encode frame (because libavcodec wants a full
+// AC3 frame for encoding, while filter input frames can be smaller or larger).
+// Return true if the
+static bool fill_buffer(af_ac3enc_t *s)
{
- struct mp_audio *out = af->data;
- af_ac3enc_t *s = af->priv;
- int num_frames = (audio->samples + mp_audio_buffer_samples(s->pending))
- / s->in_samples;
-
- int max_out_samples = s->out_samples * num_frames;
- mp_audio_realloc_min(out, max_out_samples);
- out->samples = 0;
-
- while (audio->samples > 0) {
- int ret;
-
- int consumed_pending = 0;
- struct mp_audio in_frame;
- int pending = mp_audio_buffer_samples(s->pending);
- if (pending == 0 && audio->samples >= s->in_samples) {
- in_frame = *audio;
- mp_audio_skip_samples(audio, s->in_samples);
- } else {
- if (pending > 0 && pending < s->in_samples) {
- struct mp_audio tmp = *audio;
- tmp.samples = MPMIN(tmp.samples, s->in_samples);
- mp_audio_buffer_append(s->pending, &tmp);
- mp_audio_skip_samples(audio, tmp.samples);
- }
- mp_audio_buffer_peek(s->pending, &in_frame);
- if (in_frame.samples < s->in_samples)
- break;
- consumed_pending = s->in_samples;
- }
- in_frame.samples = s->in_samples;
+ if (s->pending) {
+ int copy = MPMIN(s->in_samples - s->input->samples, s->pending->samples);
+ s->input->samples += copy;
+ mp_audio_copy(s->input, s->input->samples - copy, s->pending, 0, copy);
+ mp_audio_skip_samples(s->pending, copy);
+ }
+ return s->input->samples >= s->in_samples;
+}
- AVFrame *frame = av_frame_alloc();
- if (!frame) {
- MP_FATAL(af, "Could not allocate memory \n");
- return -1;
- }
- frame->nb_samples = s->in_samples;
- frame->format = s->lavc_actx->sample_fmt;
- frame->channel_layout = s->lavc_actx->channel_layout;
- assert(in_frame.num_planes <= AV_NUM_DATA_POINTERS);
- frame->extended_data = frame->data;
- for (int n = 0; n < in_frame.num_planes; n++)
- frame->data[n] = in_frame.planes[n];
- frame->linesize[0] = s->in_samples * audio->sstride;
-
- int ok;
- ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
- av_frame_free(&frame);
- if (ret < 0 || !ok) {
- MP_FATAL(af, "Encode failed.\n");
- return -1;
- }
+static int filter_out(struct af_instance *af)
+{
+ af_ac3enc_t *s = af->priv;
+ if (!fill_buffer(s))
+ return 0; // need more input
- mp_audio_buffer_skip(s->pending, consumed_pending);
+ AVFrame *frame = av_frame_alloc();
+ if (!frame) {
+ MP_FATAL(af, "Could not allocate memory \n");
+ return -1;
+ }
+ frame->nb_samples = s->in_samples;
+ frame->format = s->lavc_actx->sample_fmt;
+ frame->channel_layout = s->lavc_actx->channel_layout;
+ assert(s->input->num_planes <= AV_NUM_DATA_POINTERS);
+ frame->extended_data = frame->data;
+ for (int n = 0; n < s->input->num_planes; n++)
+ frame->data[n] = s->input->planes[n];
+ frame->linesize[0] = s->input->samples * s->input->sstride;
+
+ int ok;
+ int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
+ av_frame_free(&frame);
+ s->input->samples = 0;
+ if (lavc_ret < 0 || !ok) {
+ MP_FATAL(af, "Encode failed.\n");
+ return -1;
+ }
- MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
- s->pkt.size, mp_audio_buffer_samples(s->pending));
+ MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
+ s->pkt.size, s->pending->samples);
- int frame_size = s->pkt.size;
- int header_len = 0;
- char hdr[8];
+ struct mp_audio *out =
+ mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
+ if (!out)
+ return -1;
+ mp_audio_copy_attributes(out, s->pending);
- if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
- int bsmod = s->pkt.data[5] & 0x7;
- int len = frame_size;
+ int frame_size = s->pkt.size;
+ int header_len = 0;
+ char hdr[8];
- frame_size = AC3_FRAME_SIZE * 2 * 2;
- header_len = 8;
+ if (s->cfg_add_iec61937_header && s->pkt.size > 5) {
+ int bsmod = s->pkt.data[5] & 0x7;
+ int len = frame_size;
- AV_WL16(hdr, 0xF872); // iec 61937 syncword 1
- AV_WL16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
- hdr[5] = bsmod; // bsmod
- hdr[4] = 0x01; // data-type ac3
- AV_WL16(hdr + 6, len << 3); // number of bits in payload
- }
+ frame_size = AC3_FRAME_SIZE * 2 * 2;
+ header_len = 8;
- size_t max_size = (max_out_samples - out->samples) * out->sstride;
- if (frame_size > max_size)
- abort();
-
- char *buf = (char *)out->planes[0] + out->samples * out->sstride;
- memcpy(buf, hdr, header_len);
- memcpy(buf + header_len, s->pkt.data, s->pkt.size);
- memset(buf + header_len + s->pkt.size, 0,
- frame_size - (header_len + s->pkt.size));
- swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
- out->samples += frame_size / out->sstride;
+ AV_WL16(hdr, 0xF872); // iec 61937 syncword 1
+ AV_WL16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
+ hdr[5] = bsmod; // bsmod
+ hdr[4] = 0x01; // data-type ac3
+ AV_WL16(hdr + 6, len << 3); // number of bits in payload
}
- mp_audio_buffer_append(s->pending, audio);
-
- *audio = *out;
+ if (frame_size > out->samples * out->sstride)
+ abort();
+
+ char *buf = (char *)out->planes[0];
+ memcpy(buf, hdr, header_len);
+ memcpy(buf + header_len, s->pkt.data, s->pkt.size);
+ memset(buf + header_len + s->pkt.size, 0,
+ frame_size - (header_len + s->pkt.size));
+ swap_16((uint16_t *)(buf + header_len), s->pkt.size / 2);
+ out->samples = frame_size / out->sstride;
+ af_add_output_frame(af, out);
return 0;
}
@@ -260,7 +265,8 @@ static int af_open(struct af_instance* af){
af_ac3enc_t *s = af->priv;
af->control=control;
af->uninit=uninit;
- af->filter=filter;
+ af->filter_frame = filter_frame;
+ af->filter_out = filter_out;
s->lavc_acodec = avcodec_find_encoder_by_name("ac3");
if (!s->lavc_acodec) {
@@ -291,7 +297,7 @@ static int af_open(struct af_instance* af){
av_init_packet(&s->pkt);
- s->pending = mp_audio_buffer_create(af);
+ s->input = talloc_zero(s, struct mp_audio);
if (s->cfg_bit_rate) {
int i;