X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fresample.c;h=122b725d39b1c28becbccdc5a8c69193aa5c5675;hb=4f313925ec6dd32ed9e741fca40a0e22ca0c7879;hp=76521a75e0fb7672b4101f85a68c70dc617a1683;hpb=5eac5f29e3677e9ba260c3acf8197cab540bbcde;p=ffmpeg diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 76521a75e0f..122b725d39b 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -1,6 +1,6 @@ /* * samplerate conversion for both audio and video - * Copyright (c) 2000 Fabrice Bellard. + * Copyright (c) 2000 Fabrice Bellard * * This file is part of FFmpeg. * @@ -20,14 +20,24 @@ */ /** - * @file resample.c + * @file libavcodec/resample.c * samplerate conversion for both audio and video */ #include "avcodec.h" +#include "audioconvert.h" +#include "opt.h" struct AVResampleContext; +static const char *context_to_name(void *ptr) +{ + return "audioresample"; +} + +static const AVOption options[] = {{NULL}}; +static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options }; + struct ReSampleContext { struct AVResampleContext *resample_context; short *temp[2]; @@ -35,6 +45,11 @@ struct ReSampleContext { float ratio; /* channel convert */ int input_channels, output_channels, filter_channels; + AVAudioConvert *convert_ctx[2]; + enum SampleFormat 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 }; /* n1: number of samples */ @@ -126,21 +141,25 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) } } -ReSampleContext *audio_resample_init(int output_channels, int input_channels, - int output_rate, int input_rate) +ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, + int output_rate, int input_rate, + enum SampleFormat sample_fmt_out, + enum SampleFormat sample_fmt_in, + int filter_length, int log2_phase_count, + int linear, double cutoff) { ReSampleContext *s; if ( input_channels > 2) { - av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported."); + av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n"); return NULL; } s = av_mallocz(sizeof(ReSampleContext)); if (!s) { - av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context."); + av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); return NULL; } @@ -153,8 +172,36 @@ ReSampleContext *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_format(s->sample_fmt[0])>>3; + s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3; + + if (s->sample_fmt[0] != SAMPLE_FMT_S16) { + if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1, + s->sample_fmt[0], 1, NULL, 0))) { + av_log(s, AV_LOG_ERROR, + "Cannot convert %s sample format to s16 sample format\n", + avcodec_get_sample_fmt_name(s->sample_fmt[0])); + av_free(s); + return NULL; + } + } + + if (s->sample_fmt[1] != SAMPLE_FMT_S16) { + if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1, + SAMPLE_FMT_S16, 1, NULL, 0))) { + av_log(s, AV_LOG_ERROR, + "Cannot convert s16 sample format to %s sample format\n", + avcodec_get_sample_fmt_name(s->sample_fmt[1])); + av_audio_convert_free(s->convert_ctx[0]); + av_free(s); + return NULL; + } + } + /* - * ac3 output is the only case where filter_channels could be greater than 2. + * 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. */ @@ -162,11 +209,25 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels, s->filter_channels = 2; #define TAPS 16 - s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8); + s->resample_context= av_resample_init(output_rate, input_rate, + filter_length, log2_phase_count, linear, cutoff); + + *(AVClass**)s->resample_context = &audioresample_context_class; return s; } +#if LIBAVCODEC_VERSION_MAJOR < 53 +ReSampleContext *audio_resample_init(int output_channels, int input_channels, + int output_rate, int input_rate) +{ + return av_audio_resample_init(output_channels, input_channels, + output_rate, input_rate, + SAMPLE_FMT_S16, SAMPLE_FMT_S16, + TAPS, 10, 0, 0.8); +} +#endif + /* resample audio. 'nb_samples' is the number of input samples */ /* XXX: optimize it ! */ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) @@ -175,6 +236,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl short *bufin[2]; short *bufout[2]; short *buftmp2[2], *buftmp3[2]; + short *output_bak = NULL; int lenout; if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { @@ -183,17 +245,62 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl return nb_samples; } + if (s->sample_fmt[0] != SAMPLE_FMT_S16) { + int istride[1] = { s->sample_size[0] }; + int ostride[1] = { 2 }; + const void *ibuf[1] = { input }; + void *obuf[1]; + 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]); + s->buffer_size[0] = input_size; + s->buffer[0] = av_malloc(s->buffer_size[0]); + if (!s->buffer[0]) { + av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n"); + return 0; + } + } + + 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, AV_LOG_ERROR, "Audio sample format conversion failed\n"); + return 0; + } + + input = s->buffer[0]; + } + + lenout= 4*nb_samples * s->ratio + 16; + + if (s->sample_fmt[1] != SAMPLE_FMT_S16) { + output_bak = output; + + if (!s->buffer_size[1] || s->buffer_size[1] < lenout) { + av_free(s->buffer[1]); + s->buffer_size[1] = lenout; + s->buffer[1] = av_malloc(s->buffer_size[1]); + if (!s->buffer[1]) { + av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n"); + return 0; + } + } + + output = s->buffer[1]; + } + /* XXX: move those malloc to resample init code */ for(i=0; ifilter_channels; i++){ - bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); + 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; } /* make some zoom to avoid round pb */ - lenout= (int)(4*nb_samples * s->ratio) + 16; - bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); - bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); + bufout[0]= av_malloc( lenout * sizeof(short) ); + bufout[1]= av_malloc( lenout * sizeof(short) ); if (s->input_channels == 2 && s->output_channels == 1) { @@ -233,6 +340,19 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); } + if (s->sample_fmt[1] != SAMPLE_FMT_S16) { + int istride[1] = { 2 }; + int ostride[1] = { s->sample_size[1] }; + const void *ibuf[1] = { output }; + 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, AV_LOG_ERROR, "Audio sample format convertion failed\n"); + return 0; + } + } + for(i=0; ifilter_channels; i++) av_free(bufin[i]); @@ -246,5 +366,9 @@ void audio_resample_close(ReSampleContext *s) av_resample_close(s->resample_context); av_freep(&s->temp[0]); av_freep(&s->temp[1]); + av_freep(&s->buffer[0]); + av_freep(&s->buffer[1]); + av_audio_convert_free(s->convert_ctx[0]); + av_audio_convert_free(s->convert_ctx[1]); av_free(s); }