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ffplay: dynamically allocate audio buffer

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We simply remove the fixed length VideoState->audio_buf2 and use
the previously unused VideoState->audio_buf1.

Fixes ticket #2191.

Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint 2013-01-26 22:32:39 +01:00
parent b45a3e167f
commit 571ef42dd4
1 changed files with 9 additions and 4 deletions
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13
ffplay.c
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@ -181,11 +181,11 @@ typedef struct VideoState {
AVStream *audio_st;
PacketQueue audioq;
int audio_hw_buf_size;
DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE];
uint8_t *audio_buf;
uint8_t *audio_buf1;
unsigned int audio_buf_size; /* in bytes */
unsigned int audio_buf1_size;
int audio_buf_index; /* in bytes */
int audio_write_buf_size;
AVPacket audio_pkt_temp;
@ -2143,8 +2143,9 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
if (is->swr_ctx) {
const uint8_t **in = (const uint8_t **)is->frame->extended_data;
uint8_t *out[] = {is->audio_buf2};
int out_count = sizeof(is->audio_buf2) / is->audio_tgt.channels / av_get_bytes_per_sample(is->audio_tgt.fmt);
uint8_t **out = &is->audio_buf1;
int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate + 256;
int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
if (wanted_nb_samples != is->frame->nb_samples) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt.freq / is->frame->sample_rate,
wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate) < 0) {
@ -2152,6 +2153,9 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
break;
}
}
av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
if (!is->audio_buf1)
return AVERROR(ENOMEM);
len2 = swr_convert(is->swr_ctx, out, out_count, in, is->frame->nb_samples);
if (len2 < 0) {
fprintf(stderr, "swr_convert() failed\n");
@ -2161,7 +2165,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
fprintf(stderr, "warning: audio buffer is probably too small\n");
swr_init(is->swr_ctx);
}
is->audio_buf = is->audio_buf2;
is->audio_buf = is->audio_buf1;
resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
} else {
is->audio_buf = is->frame->data[0];
@ -2437,6 +2441,7 @@ static void stream_component_close(VideoState *is, int stream_index)
av_free_packet(&is->audio_pkt);
swr_free(&is->swr_ctx);
av_freep(&is->audio_buf1);
is->audio_buf1_size = 0;
is->audio_buf = NULL;
avcodec_free_frame(&is->frame);