[FFmpeg-cvslog] avplay: use libavresample for sample format conversion and channel mixing
Justin Ruggles
git at videolan.org
Tue May 1 22:19:19 CEST 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Wed Apr 25 16:46:22 2012 -0400| [f1ffb01ee9fd3a15c395c3cf6ff362ac5cd668d0] | committer: Justin Ruggles
avplay: use libavresample for sample format conversion and channel mixing
SDL only supports s16 sample format and a limited number of channel layouts.
Some versions of SDL on some systems support 4-channel and 6-channel output,
but it's safer overall to downmix any layout with more than 2 channels to
stereo.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f1ffb01ee9fd3a15c395c3cf6ff362ac5cd668d0
---
avplay.c | 124 +++++++++++++++++++++++++++++++++++++++++++-------------------
1 files changed, 86 insertions(+), 38 deletions(-)
diff --git a/avplay.c b/avplay.c
index 0d72f46..e18b2e4 100644
--- a/avplay.c
+++ b/avplay.c
@@ -34,7 +34,7 @@
#include "libavformat/avformat.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
-#include "libavcodec/audioconvert.h"
+#include "libavresample/avresample.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
@@ -159,8 +159,12 @@ typedef struct VideoState {
int audio_buf_index; /* in bytes */
AVPacket audio_pkt_temp;
AVPacket audio_pkt;
- enum AVSampleFormat audio_src_fmt;
- AVAudioConvert *reformat_ctx;
+ enum AVSampleFormat sdl_sample_fmt;
+ uint64_t sdl_channel_layout;
+ int sdl_channels;
+ enum AVSampleFormat resample_sample_fmt;
+ uint64_t resample_channel_layout;
+ AVAudioResampleContext *avr;
AVFrame *frame;
int show_audio; /* if true, display audio samples */
@@ -743,7 +747,7 @@ static void video_audio_display(VideoState *s)
nb_freq = 1 << (rdft_bits - 1);
/* compute display index : center on currently output samples */
- channels = s->audio_st->codec->channels;
+ channels = s->sdl_channels;
nb_display_channels = channels;
if (!s->paused) {
int data_used = s->show_audio == 1 ? s->width : (2 * nb_freq);
@@ -957,8 +961,8 @@ static double get_audio_clock(VideoState *is)
hw_buf_size = audio_write_get_buf_size(is);
bytes_per_sec = 0;
if (is->audio_st) {
- bytes_per_sec = is->audio_st->codec->sample_rate *
- 2 * is->audio_st->codec->channels;
+ bytes_per_sec = is->audio_st->codec->sample_rate * is->sdl_channels *
+ av_get_bytes_per_sample(is->sdl_sample_fmt);
}
if (bytes_per_sec)
pts -= (double)hw_buf_size / bytes_per_sec;
@@ -1937,7 +1941,7 @@ static int synchronize_audio(VideoState *is, short *samples,
int n, samples_size;
double ref_clock;
- n = 2 * is->audio_st->codec->channels;
+ n = is->sdl_channels * av_get_bytes_per_sample(is->sdl_sample_fmt);
samples_size = samples_size1;
/* if not master, then we try to remove or add samples to correct the clock */
@@ -2018,6 +2022,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
for (;;) {
/* NOTE: the audio packet can contain several frames */
while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) {
+ int resample_changed, audio_resample;
+
if (!is->frame) {
if (!(is->frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
@@ -2047,39 +2053,67 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
is->frame->nb_samples,
dec->sample_fmt, 1);
- if (dec->sample_fmt != is->audio_src_fmt) {
- if (is->reformat_ctx)
- av_audio_convert_free(is->reformat_ctx);
- is->reformat_ctx= av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
- dec->sample_fmt, 1, NULL, 0);
- if (!is->reformat_ctx) {
- fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
- av_get_sample_fmt_name(dec->sample_fmt),
- av_get_sample_fmt_name(AV_SAMPLE_FMT_S16));
+ audio_resample = dec->sample_fmt != is->sdl_sample_fmt ||
+ dec->channel_layout != is->sdl_channel_layout;
+
+ resample_changed = dec->sample_fmt != is->resample_sample_fmt ||
+ dec->channel_layout != is->resample_channel_layout;
+
+ if ((!is->avr && audio_resample) || resample_changed) {
+ if (is->avr)
+ avresample_close(is->avr);
+ else if (audio_resample) {
+ int ret;
+ is->avr = avresample_alloc_context();
+ if (!is->avr) {
+ fprintf(stderr, "error allocating AVAudioResampleContext\n");
break;
+ }
+ av_opt_set_int(is->avr, "in_channel_layout", dec->channel_layout, 0);
+ av_opt_set_int(is->avr, "in_sample_fmt", dec->sample_fmt, 0);
+ av_opt_set_int(is->avr, "in_sample_rate", dec->sample_rate, 0);
+ av_opt_set_int(is->avr, "out_channel_layout", is->sdl_channel_layout, 0);
+ av_opt_set_int(is->avr, "out_sample_fmt", is->sdl_sample_fmt, 0);
+ av_opt_set_int(is->avr, "out_sample_rate", dec->sample_rate, 0);
+ if (av_get_bytes_per_sample(dec->sample_fmt) <= 2)
+ av_opt_set_int(is->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
+
+ if ((ret = avresample_open(is->avr)) < 0) {
+ fprintf(stderr, "error initializing libavresample\n");
+ break;
+ }
}
- is->audio_src_fmt= dec->sample_fmt;
+ is->resample_sample_fmt = dec->sample_fmt;
+ is->resample_channel_layout = dec->channel_layout;
}
- if (is->reformat_ctx) {
- const void *ibuf[6] = { is->frame->data[0] };
- void *obuf[6];
- int istride[6] = { av_get_bytes_per_sample(dec->sample_fmt) };
- int ostride[6] = { 2 };
- int len= data_size/istride[0];
- obuf[0] = av_realloc(is->audio_buf1, FFALIGN(len * ostride[0], 32));
- if (!obuf[0]) {
+ if (audio_resample) {
+ void *tmp_out;
+ int out_samples, out_size, out_linesize;
+ int osize = av_get_bytes_per_sample(is->sdl_sample_fmt);
+ int nb_samples = is->frame->nb_samples;
+
+ out_size = av_samples_get_buffer_size(&out_linesize,
+ is->sdl_channels,
+ nb_samples,
+ is->sdl_sample_fmt, 0);
+ tmp_out = av_realloc(is->audio_buf1, out_size);
+ if (!tmp_out)
return AVERROR(ENOMEM);
- }
- is->audio_buf1 = obuf[0];
- if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len) < 0) {
- printf("av_audio_convert() failed\n");
+ is->audio_buf1 = tmp_out;
+
+ out_samples = avresample_convert(is->avr,
+ (void **)&is->audio_buf1,
+ out_linesize, nb_samples,
+ (void **)is->frame->data,
+ is->frame->linesize[0],
+ is->frame->nb_samples);
+ if (out_samples < 0) {
+ fprintf(stderr, "avresample_convert() failed\n");
break;
}
is->audio_buf = is->audio_buf1;
- /* FIXME: existing code assume that data_size equals framesize*channels*2
- remove this legacy cruft */
- data_size = len * 2;
+ data_size = out_samples * osize * is->sdl_channels;
} else {
is->audio_buf = is->frame->data[0];
}
@@ -2087,7 +2121,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
/* if no pts, then compute it */
pts = is->audio_clock;
*pts_ptr = pts;
- n = 2 * dec->channels;
+ n = is->sdl_channels * av_get_bytes_per_sample(is->sdl_sample_fmt);
is->audio_clock += (double)data_size /
(double)(n * dec->sample_rate);
#ifdef DEBUG
@@ -2206,7 +2240,20 @@ static int stream_component_open(VideoState *is, int stream_index)
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
wanted_spec.freq = avctx->sample_rate;
wanted_spec.format = AUDIO_S16SYS;
- wanted_spec.channels = avctx->channels;
+
+ if (!avctx->channel_layout)
+ avctx->channel_layout = av_get_default_channel_layout(avctx->channels);
+ if (!avctx->channel_layout) {
+ fprintf(stderr, "unable to guess channel layout\n");
+ return -1;
+ }
+ if (avctx->channels == 1)
+ is->sdl_channel_layout = AV_CH_LAYOUT_MONO;
+ else
+ is->sdl_channel_layout = AV_CH_LAYOUT_STEREO;
+ is->sdl_channels = av_get_channel_layout_nb_channels(is->sdl_channel_layout);
+
+ wanted_spec.channels = is->sdl_channels;
wanted_spec.silence = 0;
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
wanted_spec.callback = sdl_audio_callback;
@@ -2216,7 +2263,9 @@ static int stream_component_open(VideoState *is, int stream_index)
return -1;
}
is->audio_hw_buf_size = spec.size;
- is->audio_src_fmt = AV_SAMPLE_FMT_S16;
+ is->sdl_sample_fmt = AV_SAMPLE_FMT_S16;
+ is->resample_sample_fmt = is->sdl_sample_fmt;
+ is->resample_channel_layout = is->sdl_channel_layout;
}
ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
@@ -2275,9 +2324,8 @@ static void stream_component_close(VideoState *is, int stream_index)
packet_queue_end(&is->audioq);
av_free_packet(&is->audio_pkt);
- if (is->reformat_ctx)
- av_audio_convert_free(is->reformat_ctx);
- is->reformat_ctx = NULL;
+ if (is->avr)
+ avresample_free(&is->avr);
av_freep(&is->audio_buf1);
is->audio_buf = NULL;
av_freep(&is->frame);
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