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