From 571ef42dd4eb260b213464ed15d288a887b6679a Mon Sep 17 00:00:00 2001 From: Marton Balint Date: Sat, 26 Jan 2013 22:32:39 +0100 Subject: [PATCH] ffplay: dynamically allocate audio buffer 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 --- ffplay.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/ffplay.c b/ffplay.c index 93090ad60b..96a2ee1304 100644 --- a/ffplay.c +++ b/ffplay.c @@ -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);