#include "libavcodec/audioconvert.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
+#include "libswresample/swresample.h"
#if CONFIG_AVFILTER
# include "libavfilter/avcodec.h"
PacketQueue audioq;
int audio_hw_buf_size;
/* samples output by the codec. we reserve more space for avsync
- compensation */
- DECLARE_ALIGNED(16,uint8_t,audio_buf1)[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
- DECLARE_ALIGNED(16,uint8_t,audio_buf2)[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
+ compensation, resampling and format conversion */
+ DECLARE_ALIGNED(16,uint8_t,audio_buf1)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
+ DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
uint8_t *audio_buf;
unsigned int audio_buf_size; /* in bytes */
int audio_buf_index; /* in bytes */
AVPacket audio_pkt_temp;
AVPacket audio_pkt;
enum AVSampleFormat audio_src_fmt;
- AVAudioConvert *reformat_ctx;
+ enum AVSampleFormat audio_tgt_fmt;
+ int audio_src_channels;
+ int audio_tgt_channels;
+ int64_t audio_src_channel_layout;
+ int64_t audio_tgt_channel_layout;
+ int audio_src_freq;
+ int audio_tgt_freq;
+ struct SwrContext *swr_ctx;
double audio_current_pts;
double audio_current_pts_drift;
static int loop=1;
static int framedrop=-1;
static enum ShowMode show_mode = SHOW_MODE_NONE;
+static const char *audio_codec_name;
+static const char *subtitle_codec_name;
+static const char *video_codec_name;
static int rdftspeed=20;
#if CONFIG_AVFILTER
nb_freq= 1<<(rdft_bits-1);
/* compute display index : center on currently output samples */
- channels = s->audio_st->codec->channels;
+ channels = s->audio_tgt_channels;
nb_display_channels = channels;
if (!s->paused) {
int data_used= s->show_mode == SHOW_MODE_WAVES ? s->width : (2*nb_freq);
the last buffer computation */
if (audio_callback_time) {
time_diff = av_gettime() - audio_callback_time;
- delay -= (time_diff * s->audio_st->codec->sample_rate) / 1000000;
+ delay -= (time_diff * s->audio_tgt_freq) / 1000000;
}
delay += 2*data_used;
int n, samples_size;
double ref_clock;
- n = 2 * is->audio_st->codec->channels;
+ n = av_get_bytes_per_sample(is->audio_tgt_fmt) * is->audio_tgt_channels;
samples_size = samples_size1;
/* if not master, then we try to remove or add samples to correct the clock */
avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);
if (fabs(avg_diff) >= is->audio_diff_threshold) {
- wanted_size = samples_size + ((int)(diff * is->audio_st->codec->sample_rate) * n);
+ wanted_size = samples_size + ((int)(diff * is->audio_tgt_freq) * n);
nb_samples = samples_size / n;
min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
if (wanted_size < min_size)
wanted_size = min_size;
- else if (wanted_size > max_size)
- wanted_size = max_size;
+ else if (wanted_size > FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2)))
+ wanted_size = FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2));
/* add or remove samples to correction the synchro */
if (wanted_size < samples_size) {
AVPacket *pkt_temp = &is->audio_pkt_temp;
AVPacket *pkt = &is->audio_pkt;
AVCodecContext *dec= is->audio_st->codec;
- int n, len1, data_size;
+ int len1, len2, data_size, resampled_data_size;
+ int64_t dec_channel_layout;
double pts;
int new_packet = 0;
int flush_complete = 0;
continue;
}
- 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",
+ dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels);
+
+ if (dec->sample_fmt != is->audio_src_fmt || dec_channel_layout != is->audio_src_channel_layout || dec->sample_rate != is->audio_src_freq) {
+ if (is->swr_ctx)
+ swr_free(&is->swr_ctx);
+ is->swr_ctx = swr_alloc2(NULL, is->audio_tgt_channel_layout, is->audio_tgt_fmt, is->audio_tgt_freq,
+ dec_channel_layout, dec->sample_fmt, dec->sample_rate,
+ 0, NULL);
+ if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
+ fprintf(stderr, "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
+ dec->sample_rate,
av_get_sample_fmt_name(dec->sample_fmt),
- av_get_sample_fmt_name(AV_SAMPLE_FMT_S16));
- break;
+ dec->channels,
+ is->audio_tgt_freq,
+ av_get_sample_fmt_name(is->audio_tgt_fmt),
+ is->audio_tgt_channels);
+ break;
}
- is->audio_src_fmt= dec->sample_fmt;
+ is->audio_src_channel_layout = dec_channel_layout;
+ is->audio_src_channels = dec->channels;
+ is->audio_src_freq = dec->sample_rate;
+ is->audio_src_fmt = dec->sample_fmt;
}
- if (is->reformat_ctx) {
- const void *ibuf[6]= {is->audio_buf1};
- void *obuf[6]= {is->audio_buf2};
- int istride[6]= {av_get_bytes_per_sample(dec->sample_fmt)};
- int ostride[6]= {2};
- int len= data_size/istride[0];
- if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
- printf("av_audio_convert() failed\n");
+ resampled_data_size = data_size;
+ if (is->swr_ctx) {
+ const uint8_t *in[] = {is->audio_buf1};
+ uint8_t *out[] = {is->audio_buf2};
+ len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt),
+ in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt));
+ if (len2 < 0) {
+ fprintf(stderr, "audio_resample() failed\n");
break;
}
- is->audio_buf= is->audio_buf2;
- /* FIXME: existing code assume that data_size equals framesize*channels*2
- remove this legacy cruft */
- data_size= len*2;
- }else{
+ if (len2 == sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt)) {
+ fprintf(stderr, "warning: audio buffer is probably too small\n");
+ swr_init(is->swr_ctx);
+ }
+ is->audio_buf = is->audio_buf2;
+ resampled_data_size = len2 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
+ } else {
is->audio_buf= is->audio_buf1;
}
/* if no pts, then compute it */
pts = is->audio_clock;
*pts_ptr = pts;
- n = 2 * dec->channels;
- is->audio_clock += (double)data_size /
- (double)(n * dec->sample_rate);
+ is->audio_clock += (double)data_size / (dec->channels * dec->sample_rate * av_get_bytes_per_sample(dec->sample_fmt));
#ifdef DEBUG
{
static double last_clock;
last_clock = is->audio_clock;
}
#endif
- return data_size;
+ return resampled_data_size;
}
/* free the current packet */
if (audio_size < 0) {
/* if error, just output silence */
is->audio_buf = is->audio_buf1;
- is->audio_buf_size = 1024;
+ is->audio_buf_size = 256 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
memset(is->audio_buf, 0, is->audio_buf_size);
} else {
if (is->show_mode != SHOW_MODE_VIDEO)
stream += len1;
is->audio_buf_index += len1;
}
- bytes_per_sec = is->audio_st->codec->sample_rate *
- 2 * is->audio_st->codec->channels;
+ bytes_per_sec = is->audio_tgt_freq * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
/* Let's assume the audio driver that is used by SDL has two periods. */
is->audio_current_pts = is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec;
SDL_AudioSpec wanted_spec, spec;
AVDictionary *opts;
AVDictionaryEntry *t = NULL;
+ int64_t wanted_channel_layout = 0;
if (stream_index < 0 || stream_index >= ic->nb_streams)
return -1;
opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index]);
- /* prepare audio output */
- if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
- if (avctx->channels > 0) {
- avctx->request_channels = FFMIN(2, avctx->channels);
- } else {
- avctx->request_channels = 2;
- }
- }
-
codec = avcodec_find_decoder(avctx->codec_id);
+ switch(avctx->codec_type){
+ case AVMEDIA_TYPE_AUDIO : if(audio_codec_name ) codec= avcodec_find_decoder_by_name( audio_codec_name); break;
+ case AVMEDIA_TYPE_SUBTITLE: if(subtitle_codec_name) codec= avcodec_find_decoder_by_name(subtitle_codec_name); break;
+ case AVMEDIA_TYPE_VIDEO : if(video_codec_name ) codec= avcodec_find_decoder_by_name( video_codec_name); break;
+ }
if (!codec)
return -1;
if(codec->capabilities & CODEC_CAP_DR1)
avctx->flags |= CODEC_FLAG_EMU_EDGE;
- wanted_spec.freq = avctx->sample_rate;
- wanted_spec.channels = avctx->channels;
+ if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+ wanted_channel_layout = (avctx->channel_layout && avctx->channels == av_get_channel_layout_nb_channels(avctx->channels)) ? avctx->channel_layout : av_get_default_channel_layout(avctx->channels);
+ wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
+ wanted_spec.channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
+ wanted_spec.freq = avctx->sample_rate;
+ if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
+ fprintf(stderr, "Invalid sample rate or channel count!\n");
+ return -1;
+ }
+ }
+
if (!codec ||
avcodec_open2(avctx, codec, &opts) < 0)
return -1;
/* prepare audio output */
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
- if(avctx->sample_rate <= 0 || avctx->channels <= 0){
- fprintf(stderr, "Invalid sample rate or channel count\n");
- return -1;
- }
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.silence = 0;
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
return -1;
}
is->audio_hw_buf_size = spec.size;
- is->audio_src_fmt= AV_SAMPLE_FMT_S16;
+ if (spec.format != AUDIO_S16SYS) {
+ fprintf(stderr, "SDL advised audio format %d is not supported!\n", spec.format);
+ return -1;
+ }
+ if (spec.channels != wanted_spec.channels) {
+ wanted_channel_layout = av_get_default_channel_layout(spec.channels);
+ if (!wanted_channel_layout) {
+ fprintf(stderr, "SDL advised channel count %d is not supported!\n", spec.channels);
+ return -1;
+ }
+ }
+ is->audio_src_fmt = is->audio_tgt_fmt = AV_SAMPLE_FMT_S16;
+ is->audio_src_freq = is->audio_tgt_freq = spec.freq;
+ is->audio_src_channel_layout = is->audio_tgt_channel_layout = wanted_channel_layout;
+ is->audio_src_channels = is->audio_tgt_channels = spec.channels;
}
ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
is->audio_diff_avg_count = 0;
/* since we do not have a precise anough audio fifo fullness,
we correct audio sync only if larger than this threshold */
- is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / avctx->sample_rate;
+ is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / wanted_spec.freq;
memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
packet_queue_init(&is->audioq);
SDL_CloseAudio();
packet_queue_end(&is->audioq);
- if (is->reformat_ctx)
- av_audio_convert_free(is->reformat_ctx);
- is->reformat_ctx = NULL;
+ if (is->swr_ctx)
+ swr_free(&is->swr_ctx);
break;
case AVMEDIA_TYPE_VIDEO:
packet_queue_abort(&is->videoq);
if(genpts)
ic->flags |= AVFMT_FLAG_GENPTS;
+ av_dict_set(&codec_opts, "request_channels", "2", 0);
+
opts = setup_find_stream_info_opts(ic, codec_opts);
orig_nb_streams = ic->nb_streams;
input_filename = filename;
}
+static int opt_codec(void *o, const char *opt, const char *arg)
+{
+ switch(opt[strlen(opt)-1]){
+ case 'a' : audio_codec_name = arg; break;
+ case 's' : subtitle_codec_name = arg; break;
+ case 'v' : video_codec_name = arg; break;
+ }
+ return 0;
+}
+
static int dummy;
static const OptionDef options[] = {
{ "showmode", HAS_ARG, {(void*)opt_show_mode}, "select show mode (0 = video, 1 = waves, 2 = RDFT)", "mode" },
{ "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {(void*)opt_default}, "generic catch all option", "" },
{ "i", OPT_BOOL, {(void *)&dummy}, "read specified file", "input_file"},
+ { "codec", HAS_ARG | OPT_FUNC2, {(void*)opt_codec}, "force decoder", "decoder" },
{ NULL, },
};
VideoState *is;
av_log_set_flags(AV_LOG_SKIP_REPEATED);
+ parse_loglevel(argc, argv, options);
/* register all codecs, demux and protocols */
avcodec_register_all();
projects / ffmpeg / blobdiff