X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fresample.c;h=ba8ce8910c082b3ccb7073e5bdc69f08a03735f5;hb=615b2a2cf5e93bc8701944b136891973aab84579;hp=2185a11ad7bf979dd0e33346758ee1b9b77c4e59;hpb=9b4f1cdb6a7e838c0261109e8a6262e78657ee7b;p=ffmpeg diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 2185a11ad7b..ba8ce8910c0 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -29,6 +29,8 @@ #include "libavutil/opt.h" #include "libavutil/samplefmt.h" +#define MAX_CHANNELS 8 + struct AVResampleContext; static const char *context_to_name(void *ptr) @@ -37,20 +39,22 @@ static const char *context_to_name(void *ptr) } static const AVOption options[] = {{NULL}}; -static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT }; +static const AVClass audioresample_context_class = { + "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT +}; struct ReSampleContext { struct AVResampleContext *resample_context; - short *temp[2]; + short *temp[MAX_CHANNELS]; int temp_len; float ratio; /* channel convert */ int input_channels, output_channels, filter_channels; AVAudioConvert *convert_ctx[2]; enum AVSampleFormat sample_fmt[2]; ///< input and output sample format - unsigned sample_size[2]; ///< size of one sample in sample_fmt - short *buffer[2]; ///< buffers used for conversion to S16 - unsigned buffer_size[2]; ///< sizes of allocated buffers + unsigned sample_size[2]; ///< size of one sample in sample_fmt + short *buffer[2]; ///< buffers used for conversion to S16 + unsigned buffer_size[2]; ///< sizes of allocated buffers }; /* n1: number of samples */ @@ -104,41 +108,42 @@ static void mono_to_stereo(short *output, short *input, int n1) } } -/* XXX: should use more abstract 'N' channels system */ -static void stereo_split(short *output1, short *output2, short *input, int n) +static void deinterleave(short **output, short *input, int channels, int samples) { - int i; + int i, j; - for(i=0;i 2) - { - av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n"); + if (input_channels > MAX_CHANNELS) { + av_log(NULL, AV_LOG_ERROR, + "Resampling with input channels greater than %d is unsupported.\n", + MAX_CHANNELS); + return NULL; + } + if (output_channels != input_channels && + (input_channels > 2 || + output_channels > 2 && + !(output_channels == 6 && input_channels == 2))) { + av_log(NULL, AV_LOG_ERROR, + "Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n"); return NULL; - } + } s = av_mallocz(sizeof(ReSampleContext)); - if (!s) - { + if (!s) { av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); return NULL; - } + } s->ratio = (float)output_rate / (float)input_rate; @@ -173,10 +186,10 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, if (s->output_channels < s->filter_channels) s->filter_channels = s->output_channels; - s->sample_fmt [0] = sample_fmt_in; - s->sample_fmt [1] = sample_fmt_out; - s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0])>>3; - s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1])>>3; + s->sample_fmt[0] = sample_fmt_in; + s->sample_fmt[1] = sample_fmt_out; + s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]); + s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]); if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) { if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1, @@ -201,17 +214,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, } } -/* - * AC-3 output is the only case where filter_channels could be greater than 2. - * input channels can't be greater than 2, so resample the 2 channels and then - * expand to 6 channels after the resampling. - */ - if(s->filter_channels>2) - s->filter_channels = 2; - -#define TAPS 16 - s->resample_context= av_resample_init(output_rate, input_rate, - filter_length, log2_phase_count, linear, cutoff); + s->resample_context = av_resample_init(output_rate, input_rate, + filter_length, log2_phase_count, + linear, cutoff); *(const AVClass**)s->resample_context = &audioresample_context_class; @@ -223,9 +228,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) { int i, nb_samples1; - short *bufin[2]; - short *bufout[2]; - short *buftmp2[2], *buftmp3[2]; + short *bufin[MAX_CHANNELS]; + short *bufout[MAX_CHANNELS]; + short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS]; short *output_bak = NULL; int lenout; @@ -240,7 +245,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl int ostride[1] = { 2 }; const void *ibuf[1] = { input }; void *obuf[1]; - unsigned input_size = nb_samples*s->input_channels*2; + unsigned input_size = nb_samples * s->input_channels * 2; if (!s->buffer_size[0] || s->buffer_size[0] < input_size) { av_free(s->buffer[0]); @@ -255,15 +260,16 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl obuf[0] = s->buffer[0]; if (av_audio_convert(s->convert_ctx[0], obuf, ostride, - ibuf, istride, nb_samples*s->input_channels) < 0) { - av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n"); + ibuf, istride, nb_samples * s->input_channels) < 0) { + av_log(s->resample_context, AV_LOG_ERROR, + "Audio sample format conversion failed\n"); return 0; } - input = s->buffer[0]; + input = s->buffer[0]; } - lenout= 4*nb_samples * s->ratio + 16; + lenout = 4 * nb_samples * s->ratio + 16; if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { output_bak = output; @@ -282,52 +288,50 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl } /* XXX: move those malloc to resample init code */ - for(i=0; ifilter_channels; i++){ - bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); + for (i = 0; i < s->filter_channels; i++) { + bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short)); memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); buftmp2[i] = bufin[i] + s->temp_len; + bufout[i] = av_malloc(lenout * sizeof(short)); } - /* make some zoom to avoid round pb */ - bufout[0]= av_malloc( lenout * sizeof(short) ); - bufout[1]= av_malloc( lenout * sizeof(short) ); - - if (s->input_channels == 2 && - s->output_channels == 1) { + if (s->input_channels == 2 && s->output_channels == 1) { buftmp3[0] = output; stereo_to_mono(buftmp2[0], input, nb_samples); } else if (s->output_channels >= 2 && s->input_channels == 1) { buftmp3[0] = bufout[0]; - memcpy(buftmp2[0], input, nb_samples*sizeof(short)); - } else if (s->output_channels >= 2) { - buftmp3[0] = bufout[0]; - buftmp3[1] = bufout[1]; - stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); + memcpy(buftmp2[0], input, nb_samples * sizeof(short)); + } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) { + for (i = 0; i < s->input_channels; i++) { + buftmp3[i] = bufout[i]; + } + deinterleave(buftmp2, input, s->input_channels, nb_samples); } else { buftmp3[0] = output; - memcpy(buftmp2[0], input, nb_samples*sizeof(short)); + memcpy(buftmp2[0], input, nb_samples * sizeof(short)); } nb_samples += s->temp_len; /* resample each channel */ nb_samples1 = 0; /* avoid warning */ - for(i=0;ifilter_channels;i++) { + for (i = 0; i < s->filter_channels; i++) { int consumed; - int is_last= i+1 == s->filter_channels; + int is_last = i + 1 == s->filter_channels; - nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last); - s->temp_len= nb_samples - consumed; - s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short)); - memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short)); + nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], + &consumed, nb_samples, lenout, is_last); + s->temp_len = nb_samples - consumed; + s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short)); + memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short)); } if (s->output_channels == 2 && s->input_channels == 1) { mono_to_stereo(output, buftmp3[0], nb_samples1); - } else if (s->output_channels == 2) { - stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); - } else if (s->output_channels == 6) { + } else if (s->output_channels == 6 && s->input_channels == 2) { ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); + } else if (s->output_channels == s->input_channels && s->input_channels >= 2) { + interleave(output, buftmp3, s->output_channels, nb_samples1); } if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { @@ -337,25 +341,27 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl void *obuf[1] = { output_bak }; if (av_audio_convert(s->convert_ctx[1], obuf, ostride, - ibuf, istride, nb_samples1*s->output_channels) < 0) { - av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n"); + ibuf, istride, nb_samples1 * s->output_channels) < 0) { + av_log(s->resample_context, AV_LOG_ERROR, + "Audio sample format convertion failed\n"); return 0; } } - for(i=0; ifilter_channels; i++) + for (i = 0; i < s->filter_channels; i++) { av_free(bufin[i]); + av_free(bufout[i]); + } - av_free(bufout[0]); - av_free(bufout[1]); return nb_samples1; } void audio_resample_close(ReSampleContext *s) { + int i; av_resample_close(s->resample_context); - av_freep(&s->temp[0]); - av_freep(&s->temp[1]); + for (i = 0; i < s->filter_channels; i++) + av_freep(&s->temp[i]); av_freep(&s->buffer[0]); av_freep(&s->buffer[1]); av_audio_convert_free(s->convert_ctx[0]);