RepoMirrors
/
ffmpeg
mirror of https://git.ffmpeg.org/ffmpeg.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity

Merge commit '8729698d50739524665090e083d1bfdf28235724' 

* commit '8729698d50739524665090e083d1bfdf28235724':
  rtsp: Recheck the reordering queue if getting a new packet
  lavr: log channel conversion description for any-to-any functions
  lavr: mix: reduce the mixing matrix when possible
  lavr: cosmetics: reindent

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2013-01-08 13:05:02 +01:00
parent 249fca3df9 8729698d50
commit 48d30f6733
3 changed files with 223 additions and 59 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
libavformat/rtsp.c
View File

@ -1863,6 +1863,7 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
rt->cur_transport_priv = NULL;
}
redo:
if (rt->transport == RTSP_TRANSPORT_RTP) {
int i;
int64_t first_queue_time = 0;
@ -1878,12 +1879,15 @@ int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
first_queue_st = rt->rtsp_streams[i];
}
}
if (first_queue_time)
if (first_queue_time) {
wait_end = first_queue_time + s->max_delay;
} else {
wait_end = 0;
first_queue_st = NULL;
}
}
/* read next RTP packet */
redo:
if (!rt->recvbuf) {
rt->recvbuf = av_malloc(RECVBUF_SIZE);
if (!rt->recvbuf)

53
libavresample/audio_convert.c
View File

@ -378,32 +378,35 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
}
}
} else {
switch (ac->func_type) {
case CONV_FUNC_TYPE_FLAT: {
if (!in->is_planar)
len *= in->channels;
if (use_generic) {
for (p = 0; p < ac->planes; p++)
ac->conv_flat_generic(out->data[p], in->data[p], len);
} else {
for (p = 0; p < ac->planes; p++)
ac->conv_flat(out->data[p], in->data[p], len);
switch (ac->func_type) {
case CONV_FUNC_TYPE_FLAT: {
if (!in->is_planar)
len *= in->channels;
if (use_generic) {
for (p = 0; p < ac->planes; p++)
ac->conv_flat_generic(out->data[p], in->data[p], len);
} else {
for (p = 0; p < ac->planes; p++)
ac->conv_flat(out->data[p], in->data[p], len);
}
break;
}
case CONV_FUNC_TYPE_INTERLEAVE:
if (use_generic)
ac->conv_interleave_generic(out->data[0], in->data, len,
ac->channels);
else
ac->conv_interleave(out->data[0], in->data, len, ac->channels);
break;
case CONV_FUNC_TYPE_DEINTERLEAVE:
if (use_generic)
ac->conv_deinterleave_generic(out->data, in->data[0], len,
ac->channels);
else
ac->conv_deinterleave(out->data, in->data[0], len,
ac->channels);
break;
}
break;
}
case CONV_FUNC_TYPE_INTERLEAVE:
if (use_generic)
ac->conv_interleave_generic(out->data[0], in->data, len, ac->channels);
else
ac->conv_interleave(out->data[0], in->data, len, ac->channels);
break;
case CONV_FUNC_TYPE_DEINTERLEAVE:
if (use_generic)
ac->conv_deinterleave_generic(out->data, in->data[0], len, ac->channels);
else
ac->conv_deinterleave(out->data, in->data[0], len, ac->channels);
break;
}
}
out->nb_samples = in->nb_samples;

221
libavresample/audio_mix.c
View File

@ -47,6 +47,11 @@ struct AudioMix {
mix_func *mix;
mix_func *mix_generic;
int in_matrix_channels;
int out_matrix_channels;
int output_zero[AVRESAMPLE_MAX_CHANNELS];
int input_skip[AVRESAMPLE_MAX_CHANNELS];
int output_skip[AVRESAMPLE_MAX_CHANNELS];
int16_t *matrix_q8[AVRESAMPLE_MAX_CHANNELS];
int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS];
float *matrix_flt[AVRESAMPLE_MAX_CHANNELS];
@ -59,8 +64,8 @@ void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
const char *descr, void *mix_func)
{
if (fmt == am->fmt && coeff_type == am->coeff_type &&
( in_channels == am->in_channels || in_channels == 0) &&
(out_channels == am->out_channels || out_channels == 0)) {
( in_channels == am->in_matrix_channels || in_channels == 0) &&
(out_channels == am->out_matrix_channels || out_channels == 0)) {
char chan_str[16];
am->mix = mix_func;
am->func_descr = descr;
@ -82,11 +87,12 @@ void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
} else if (out_channels) {
snprintf(chan_str, sizeof(chan_str), "[any to %d] ",
out_channels);
} else {
snprintf(chan_str, sizeof(chan_str), "[any to any] ");
}
av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] "
"[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt),
coeff_type_names[coeff_type],
(in_channels || out_channels) ? chan_str : "", descr);
coeff_type_names[coeff_type], chan_str, descr);
}
}
@ -278,6 +284,12 @@ static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len,
static int mix_function_init(AudioMix *am)
{
/* no need to set a mix function when we're skipping mixing */
if (!am->in_matrix_channels || !am->out_matrix_channels) {
am->func_descr = "n/a";
return 0;
}
/* any-to-any C versions */
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
@ -379,25 +391,34 @@ AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr)
goto error;
}
ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels);
if (ret < 0) {
av_log(avr, AV_LOG_ERROR, "error setting mix matrix\n");
av_free(matrix_dbl);
goto error;
}
av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name),
avr->in_channels, avr->in_channel_layout);
av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name),
avr->out_channels, avr->out_channel_layout);
av_log(avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n",
in_layout_name, out_layout_name);
av_log(avr, AV_LOG_DEBUG, "matrix size: %d x %d\n",
am->in_matrix_channels, am->out_matrix_channels);
for (i = 0; i < avr->out_channels; i++) {
for (j = 0; j < avr->in_channels; j++) {
av_log(avr, AV_LOG_DEBUG, " %0.3f ",
matrix_dbl[i * avr->in_channels + j]);
if (am->output_zero[i])
av_log(avr, AV_LOG_DEBUG, " (ZERO)");
else if (am->input_skip[j] || am->output_skip[i])
av_log(avr, AV_LOG_DEBUG, " (SKIP)");
else
av_log(avr, AV_LOG_DEBUG, " %0.3f ",
matrix_dbl[i * avr->in_channels + j]);
}
av_log(avr, AV_LOG_DEBUG, "\n");
}
ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels);
if (ret < 0) {
av_free(matrix_dbl);
goto error;
}
av_free(matrix_dbl);
}
@ -435,6 +456,7 @@ int ff_audio_mix(AudioMix *am, AudioData *src)
{
int use_generic = 1;
int len = src->nb_samples;
int i, j;
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
@ -450,11 +472,35 @@ int ff_audio_mix(AudioMix *am, AudioData *src)
src->nb_samples, am->in_channels, am->out_channels,
use_generic ? am->func_descr_generic : am->func_descr);
if (use_generic)
am->mix_generic(src->data, am->matrix, len, am->out_channels,
am->in_channels);
else
am->mix(src->data, am->matrix, len, am->out_channels, am->in_channels);
if (am->in_matrix_channels && am->out_matrix_channels) {
uint8_t **data;
uint8_t *data0[AVRESAMPLE_MAX_CHANNELS];
if (am->out_matrix_channels < am->out_channels ||
am->in_matrix_channels < am->in_channels) {
for (i = 0, j = 0; i < FFMAX(am->in_channels, am->out_channels); i++) {
if (am->input_skip[i] || am->output_skip[i] || am->output_zero[i])
continue;
data0[j++] = src->data[i];
}
data = data0;
} else {
data = src->data;
}
if (use_generic)
am->mix_generic(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
else
am->mix(data, am->matrix, len, am->out_matrix_channels,
am->in_matrix_channels);
}
if (am->out_matrix_channels < am->out_channels) {
for (i = 0; i < am->out_channels; i++)
if (am->output_zero[i])
av_samples_set_silence(&src->data[i], 0, len, 1, am->fmt);
}
ff_audio_data_set_channels(src, am->out_channels);
@ -463,7 +509,7 @@ int ff_audio_mix(AudioMix *am, AudioData *src)
int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
{
int i, o;
int i, o, i0, o0;
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
@ -476,9 +522,19 @@ int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \
return AVERROR(EINVAL); \
} \
for (o = 0; o < am->out_channels; o++) \
for (i = 0; i < am->in_channels; i++) \
matrix[o * stride + i] = am->matrix_ ## suffix[o][i] * (scale);
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
if (am->input_skip[i] || am->output_zero[o]) \
matrix[o * stride + i] = 0.0; \
else \
matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \
(scale); \
if (!am->input_skip[i]) \
i0++; \
} \
if (!am->output_zero[o]) \
o0++; \
}
switch (am->coeff_type) {
case AV_MIX_COEFF_TYPE_Q8:
@ -500,7 +556,7 @@ int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride)
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
{
int i, o;
int i, o, i0, o0;
if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS ||
am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) {
@ -513,19 +569,123 @@ int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
am->matrix = NULL;
}
memset(am->output_zero, 0, sizeof(am->output_zero));
memset(am->input_skip, 0, sizeof(am->input_skip));
memset(am->output_skip, 0, sizeof(am->output_zero));
am->in_matrix_channels = am->in_channels;
am->out_matrix_channels = am->out_channels;
/* exclude output channels if they can be zeroed instead of mixed */
for (o = 0; o < am->out_channels; o++) {
int zero = 1;
/* check if the output is always silent */
for (i = 0; i < am->in_channels; i++) {
if (matrix[o * stride + i] != 0.0) {
zero = 0;
break;
}
}
/* check if the corresponding input channel makes a contribution to
any output channel */
if (o < am->in_channels) {
for (i = 0; i < am->out_channels; i++) {
if (matrix[i * stride + o] != 0.0) {
zero = 0;
break;
}
}
}
if (zero) {
am->output_zero[o] = 1;
am->out_matrix_channels--;
}
}
if (am->out_matrix_channels == 0) {
am->in_matrix_channels = 0;
return 0;
}
/* skip input channels that contribute fully only to the corresponding
output channel */
for (i = 0; i < FFMIN(am->in_channels, am->out_channels); i++) {
int skip = 1;
for (o = 0; o < am->out_channels; o++) {
if ((o != i && matrix[o * stride + i] != 0.0) ||
(o == i && matrix[o * stride + i] != 1.0)) {
skip = 0;
break;
}
}
if (skip) {
am->input_skip[i] = 1;
am->in_matrix_channels--;
}
}
/* skip input channels that do not contribute to any output channel */
for (; i < am->in_channels; i++) {
int contrib = 0;
for (o = 0; o < am->out_channels; o++) {
if (matrix[o * stride + i] != 0.0) {
contrib = 1;
break;
}
}
if (!contrib) {
am->input_skip[i] = 1;
am->in_matrix_channels--;
}
}
if (am->in_matrix_channels == 0) {
am->out_matrix_channels = 0;
return 0;
}
/* skip output channels that only get full contribution from the
corresponding input channel */
for (o = 0; o < FFMIN(am->in_channels, am->out_channels); o++) {
int skip = 1;
for (i = 0; i < am->in_channels; i++) {
if ((o != i && matrix[o * stride + i] != 0.0) ||
(o == i && matrix[o * stride + i] != 1.0)) {
skip = 0;
break;
}
}
if (skip) {
am->output_skip[o] = 1;
am->out_matrix_channels--;
}
}
if (am->out_matrix_channels == 0) {
am->in_matrix_channels = 0;
return 0;
}
#define CONVERT_MATRIX(type, expr) \
am->matrix_## type[0] = av_mallocz(am->out_channels * am->in_channels * \
am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \
am->in_matrix_channels * \
sizeof(*am->matrix_## type[0])); \
if (!am->matrix_## type[0]) \
return AVERROR(ENOMEM); \
for (o = 0; o < am->out_channels; o++) { \
if (o > 0) \
am->matrix_## type[o] = am->matrix_## type[o - 1] + \
am->in_channels; \
for (i = 0; i < am->in_channels; i++) { \
double v = matrix[o * stride + i]; \
am->matrix_## type[o][i] = expr; \
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
if (am->output_zero[o] || am->output_skip[o]) \
continue; \
if (o0 > 0) \
am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \
am->in_matrix_channels; \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
double v; \
if (am->input_skip[i]) \
continue; \
v = matrix[o * stride + i]; \
am->matrix_## type[o0][i0] = expr; \
i0++; \
} \
o0++; \
} \
am->matrix = (void **)am->matrix_## type;
@ -544,8 +704,5 @@ int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride)
return AVERROR(EINVAL);
}
/* TODO: detect situations where we can just swap around pointers
instead of doing matrix multiplications with 0.0 and 1.0 */
return 0;
}