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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00

lavu/audio_fifo: switch to new FIFO API

This commit is contained in:
Anton Khirnov 2022-01-07 11:41:42 +01:00 committed by Andreas Rheinhardt
parent 082136209c
commit d1bd189c63

View File

@ -32,7 +32,7 @@
#include "samplefmt.h"
struct AVAudioFifo {
AVFifoBuffer **buf; /**< single buffer for interleaved, per-channel buffers for planar */
AVFifo **buf; /**< single buffer for interleaved, per-channel buffers for planar */
int nb_buffers; /**< number of buffers */
int nb_samples; /**< number of samples currently in the FIFO */
int allocated_samples; /**< current allocated size, in samples */
@ -48,7 +48,7 @@ void av_audio_fifo_free(AVAudioFifo *af)
if (af->buf) {
int i;
for (i = 0; i < af->nb_buffers; i++) {
av_fifo_freep(&af->buf[i]);
av_fifo_freep2(&af->buf[i]);
}
av_freep(&af->buf);
}
@ -80,7 +80,7 @@ AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
goto error;
for (i = 0; i < af->nb_buffers; i++) {
af->buf[i] = av_fifo_alloc(buf_size);
af->buf[i] = av_fifo_alloc2(buf_size, 1, 0);
if (!af->buf[i])
goto error;
}
@ -95,15 +95,19 @@ error:
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
{
const size_t cur_size = av_fifo_can_read (af->buf[0]) +
av_fifo_can_write(af->buf[0]);
int i, ret, buf_size;
if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples,
af->sample_fmt, 1)) < 0)
return ret;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_realloc2(af->buf[i], buf_size)) < 0)
return ret;
if (buf_size > cur_size) {
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_grow2(af->buf[i], buf_size - cur_size)) < 0)
return ret;
}
}
af->allocated_samples = nb_samples;
return 0;
@ -126,8 +130,8 @@ int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
ret = av_fifo_generic_write(af->buf[i], data[i], size, NULL);
if (ret != size)
ret = av_fifo_write(af->buf[i], data[i], size);
if (ret < 0)
return AVERROR_BUG;
}
af->nb_samples += nb_samples;
@ -137,21 +141,7 @@ int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
{
int i, ret, size;
if (nb_samples < 0)
return AVERROR(EINVAL);
nb_samples = FFMIN(nb_samples, af->nb_samples);
if (!nb_samples)
return 0;
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_generic_peek(af->buf[i], data[i], size, NULL)) < 0)
return AVERROR_BUG;
}
return nb_samples;
return av_audio_fifo_peek_at(af, data, nb_samples, 0);
}
int av_audio_fifo_peek_at(AVAudioFifo *af, void **data, int nb_samples, int offset)
@ -171,7 +161,7 @@ int av_audio_fifo_peek_at(AVAudioFifo *af, void **data, int nb_samples, int offs
offset *= af->sample_size;
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
if ((ret = av_fifo_generic_peek_at(af->buf[i], data[i], offset, size, NULL)) < 0)
if ((ret = av_fifo_peek(af->buf[i], data[i], size, offset)) < 0)
return AVERROR_BUG;
}
@ -190,7 +180,7 @@ int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++) {
if (av_fifo_generic_read(af->buf[i], data[i], size, NULL) < 0)
if (av_fifo_read(af->buf[i], data[i], size) < 0)
return AVERROR_BUG;
}
af->nb_samples -= nb_samples;
@ -209,7 +199,7 @@ int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
if (nb_samples) {
size = nb_samples * af->sample_size;
for (i = 0; i < af->nb_buffers; i++)
av_fifo_drain(af->buf[i], size);
av_fifo_drain2(af->buf[i], size);
af->nb_samples -= nb_samples;
}
return 0;
@ -220,7 +210,7 @@ void av_audio_fifo_reset(AVAudioFifo *af)
int i;
for (i = 0; i < af->nb_buffers; i++)
av_fifo_reset(af->buf[i]);
av_fifo_reset2(af->buf[i]);
af->nb_samples = 0;
}