# HG changeset patch # User bcoudurier # Date 1234393030 0 # Node ID cbeaa8c0fe4fc50f7f5874082619981f2b1729e8 # Parent eda229beb608edf9b363393a00e3320656e41263 extend resampling API, add S16 internal conversion diff -r eda229beb608 -r cbeaa8c0fe4f avcodec.h --- a/avcodec.h Wed Feb 11 19:07:25 2009 +0000 +++ b/avcodec.h Wed Feb 11 22:57:10 2009 +0000 @@ -30,7 +30,7 @@ #include "libavutil/avutil.h" #define LIBAVCODEC_VERSION_MAJOR 52 -#define LIBAVCODEC_VERSION_MINOR 14 +#define LIBAVCODEC_VERSION_MINOR 15 #define LIBAVCODEC_VERSION_MICRO 0 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ @@ -2443,8 +2443,36 @@ typedef struct ReSampleContext ReSampleContext; -ReSampleContext *audio_resample_init(int output_channels, int input_channels, - int output_rate, int input_rate); +#if LIBAVCODEC_VERSION_MAJOR < 53 +/** + * @deprecated Use av_audio_resample_init() instead. + */ +attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, int input_channels, + int output_rate, int input_rate); +#endif +/** + * Initializes audio resampling context + * + * @param output_channels number of output channels + * @param input_channels number of input channels + * @param output_rate output sample rate + * @param input_rate input sample rate + * @param sample_fmt_out requested output sample format + * @param sample_fmt_in input sample format + * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq + * @param log2_phase_count log2 of the number of entries in the polyphase filterbank + * @param linear If 1 then the used FIR filter will be linearly interpolated + between the 2 closest, if 0 the closest will be used + * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate + * @return allocated ReSampleContext, NULL if error occured + */ +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); + int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples); void audio_resample_close(ReSampleContext *s); diff -r eda229beb608 -r cbeaa8c0fe4f resample.c --- a/resample.c Wed Feb 11 19:07:25 2009 +0000 +++ b/resample.c Wed Feb 11 22:57:10 2009 +0000 @@ -25,16 +25,32 @@ */ #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 { + const AVClass *av_class; struct AVResampleContext *resample_context; short *temp[2]; int temp_len; 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 */ @@ -126,8 +142,12 @@ } } -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; @@ -153,6 +173,34 @@ 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; + } + } + /* * 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 @@ -162,11 +210,25 @@ 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); + + s->av_class= &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 +237,7 @@ 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) { @@ -183,6 +246,52 @@ 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*s->sample_size[0]; + + 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]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); @@ -191,7 +300,6 @@ } /* make some zoom to avoid round pb */ - lenout= 4*nb_samples * s->ratio + 16; bufout[0]= av_malloc( lenout * sizeof(short) ); bufout[1]= av_malloc( lenout * sizeof(short) ); @@ -233,6 +341,19 @@ 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 +367,9 @@ 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); }