From 074a00d192c0e749d677b008b337da42597e780f Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 19 Dec 2012 14:58:57 -0500 Subject: lavr: add a public function for setting a custom channel map This allows reordering, duplication, and silencing of input channels. --- libavresample/audio_convert.c | 49 ++++++++++++- libavresample/audio_convert.h | 4 +- libavresample/audio_data.c | 32 ++++++++- libavresample/audio_data.h | 3 +- libavresample/avresample.h | 30 ++++++++ libavresample/dither.c | 25 +++++-- libavresample/dither.h | 2 +- libavresample/internal.h | 23 +++++++ libavresample/utils.c | 156 +++++++++++++++++++++++++++++++++++++++--- libavresample/version.h | 4 +- 10 files changed, 301 insertions(+), 27 deletions(-) (limited to 'libavresample') diff --git a/libavresample/audio_convert.c b/libavresample/audio_convert.c index 288f0f41f1..b57d2fa650 100644 --- a/libavresample/audio_convert.c +++ b/libavresample/audio_convert.c @@ -50,6 +50,7 @@ struct AudioConvert { DitherContext *dc; enum AVSampleFormat in_fmt; enum AVSampleFormat out_fmt; + int apply_map; int channels; int planes; int ptr_align; @@ -259,7 +260,8 @@ void ff_audio_convert_free(AudioConvert **ac) AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, - int channels, int sample_rate) + int channels, int sample_rate, + int apply_map) { AudioConvert *ac; int in_planar, out_planar; @@ -272,11 +274,13 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, ac->out_fmt = out_fmt; ac->in_fmt = in_fmt; ac->channels = channels; + ac->apply_map = apply_map; if (avr->dither_method != AV_RESAMPLE_DITHER_NONE && av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 && av_get_bytes_per_sample(in_fmt) > 2) { - ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate); + ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, + apply_map); if (!ac->dc) { av_free(ac); return NULL; @@ -309,6 +313,7 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic = 1; int len = in->nb_samples; + int p; if (ac->dc) { /* dithered conversion */ @@ -335,9 +340,46 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) av_get_sample_fmt_name(ac->out_fmt), use_generic ? ac->func_descr_generic : ac->func_descr); + if (ac->apply_map) { + ChannelMapInfo *map = &ac->avr->ch_map_info; + + if (!av_sample_fmt_is_planar(ac->out_fmt)) { + av_log(ac->avr, AV_LOG_ERROR, "cannot remap packed format during conversion\n"); + return AVERROR(EINVAL); + } + + if (map->do_remap) { + if (av_sample_fmt_is_planar(ac->in_fmt)) { + conv_func_flat *convert = use_generic ? ac->conv_flat_generic : + ac->conv_flat; + + for (p = 0; p < ac->planes; p++) + if (map->channel_map[p] >= 0) + convert(out->data[p], in->data[map->channel_map[p]], len); + } else { + uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; + conv_func_deinterleave *convert = use_generic ? + ac->conv_deinterleave_generic : + ac->conv_deinterleave; + + for (p = 0; p < ac->channels; p++) + data[map->input_map[p]] = out->data[p]; + + convert(data, in->data[0], len, ac->channels); + } + } + if (map->do_copy || map->do_zero) { + for (p = 0; p < ac->planes; p++) { + if (map->channel_copy[p]) + memcpy(out->data[p], out->data[map->channel_copy[p]], + len * out->stride); + else if (map->channel_zero[p]) + av_samples_set_silence(&out->data[p], 0, len, 1, ac->out_fmt); + } + } + } else { switch (ac->func_type) { case CONV_FUNC_TYPE_FLAT: { - int p; if (!in->is_planar) len *= in->channels; if (use_generic) { @@ -362,6 +404,7 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) ac->conv_deinterleave(out->data, in->data[0], len, ac->channels); break; } + } out->nb_samples = in->nb_samples; return 0; diff --git a/libavresample/audio_convert.h b/libavresample/audio_convert.h index 7d47b15bf3..6a3089d4fb 100644 --- a/libavresample/audio_convert.h +++ b/libavresample/audio_convert.h @@ -58,12 +58,14 @@ void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt, * @param in_fmt input sample format * @param channels number of channels * @param sample_rate sample rate (used for dithering) + * @param apply_map apply channel map during conversion * @return newly-allocated AudioConvert context */ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, - int channels, int sample_rate); + int channels, int sample_rate, + int apply_map); /** * Free AudioConvert. diff --git a/libavresample/audio_data.c b/libavresample/audio_data.c index 199a68cb11..c52f518e9a 100644 --- a/libavresample/audio_data.c +++ b/libavresample/audio_data.c @@ -213,7 +213,7 @@ void ff_audio_data_free(AudioData **a) av_freep(a); } -int ff_audio_data_copy(AudioData *dst, AudioData *src) +int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map) { int ret, p; @@ -221,6 +221,11 @@ int ff_audio_data_copy(AudioData *dst, AudioData *src) if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels) return AVERROR(EINVAL); + if (map && !src->is_planar) { + av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n"); + return AVERROR(EINVAL); + } + /* if the input is empty, just empty the output */ if (!src->nb_samples) { dst->nb_samples = 0; @@ -233,8 +238,29 @@ int ff_audio_data_copy(AudioData *dst, AudioData *src) return ret; /* copy data */ - for (p = 0; p < src->planes; p++) - memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride); + if (map) { + if (map->do_remap) { + for (p = 0; p < src->planes; p++) { + if (map->channel_map[p] >= 0) + memcpy(dst->data[p], src->data[map->channel_map[p]], + src->nb_samples * src->stride); + } + } + if (map->do_copy || map->do_zero) { + for (p = 0; p < src->planes; p++) { + if (map->channel_copy[p]) + memcpy(dst->data[p], dst->data[map->channel_copy[p]], + src->nb_samples * src->stride); + else if (map->channel_zero[p]) + av_samples_set_silence(&dst->data[p], 0, src->nb_samples, + 1, dst->sample_fmt); + } + } + } else { + for (p = 0; p < src->planes; p++) + memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride); + } + dst->nb_samples = src->nb_samples; return 0; diff --git a/libavresample/audio_data.h b/libavresample/audio_data.h index 4e53e31c55..97236bb5de 100644 --- a/libavresample/audio_data.h +++ b/libavresample/audio_data.h @@ -118,9 +118,10 @@ void ff_audio_data_free(AudioData **a); * * @param out output AudioData * @param in input AudioData + * @param map channel map, NULL if not remapping * @return 0 on success, negative AVERROR value on error */ -int ff_audio_data_copy(AudioData *out, AudioData *in); +int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map); /** * Append data from one AudioData to the end of another. diff --git a/libavresample/avresample.h b/libavresample/avresample.h index 0012787404..d26f2ca223 100644 --- a/libavresample/avresample.h +++ b/libavresample/avresample.h @@ -258,6 +258,36 @@ int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, int stride); +/** + * Set a customized input channel mapping. + * + * This function can only be called when the allocated context is not open. + * Also, the input channel layout must have already been set. + * + * Calling avresample_close() on the context will clear the channel mapping. + * + * The map for each input channel specifies the channel index in the source to + * use for that particular channel, or -1 to mute the channel. Source channels + * can be duplicated by using the same index for multiple input channels. + * + * Examples: + * + * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to Libav order (L,R,C,LFE,Ls,Rs): + * { 1, 2, 0, 5, 3, 4 } + * + * Muting the 3rd channel in 4-channel input: + * { 0, 1, -1, 3 } + * + * Duplicating the left channel of stereo input: + * { 0, 0 } + * + * @param avr audio resample context + * @param channel_map customized input channel mapping + * @return 0 on success, negative AVERROR code on failure + */ +int avresample_set_channel_mapping(AVAudioResampleContext *avr, + const int *channel_map); + /** * Set compensation for resampling. * diff --git a/libavresample/dither.c b/libavresample/dither.c index 9c1e1c1101..dfff03e756 100644 --- a/libavresample/dither.c +++ b/libavresample/dither.c @@ -53,6 +53,8 @@ typedef struct DitherState { struct DitherContext { DitherDSPContext ddsp; enum AVResampleDitherMethod method; + int apply_map; + ChannelMapInfo *ch_map_info; int mute_dither_threshold; // threshold for disabling dither int mute_reset_threshold; // threshold for resetting noise shaping @@ -251,17 +253,23 @@ int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src) return ret; } - if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) { + if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) { /* make sure flt_data is large enough for the input */ ret = ff_audio_data_realloc(c->flt_data, src->nb_samples); if (ret < 0) return ret; flt_data = c->flt_data; + } + if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) { /* convert input samples to fltp and scale to s16 range */ ret = ff_audio_convert(c->ac_in, flt_data, src); if (ret < 0) return ret; + } else if (c->apply_map) { + ret = ff_audio_data_copy(flt_data, src, c->ch_map_info); + if (ret < 0) + return ret; } else { flt_data = src; } @@ -333,7 +341,7 @@ static void dither_init(DitherDSPContext *ddsp, DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, - int channels, int sample_rate) + int channels, int sample_rate, int apply_map) { AVLFG seed_gen; DitherContext *c; @@ -350,6 +358,10 @@ DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, if (!c) return NULL; + c->apply_map = apply_map; + if (apply_map) + c->ch_map_info = &avr->ch_map_info; + if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS && sample_rate != 48000 && sample_rate != 44100) { av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz " @@ -379,19 +391,20 @@ DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, goto fail; c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P, - channels, sample_rate); + channels, sample_rate, 0); if (!c->ac_out) goto fail; } - if (in_fmt != AV_SAMPLE_FMT_FLTP) { + if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) { c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP, "dither flt buffer"); if (!c->flt_data) goto fail; - + } + if (in_fmt != AV_SAMPLE_FMT_FLTP) { c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt, - channels, sample_rate); + channels, sample_rate, c->apply_map); if (!c->ac_in) goto fail; } diff --git a/libavresample/dither.h b/libavresample/dither.h index 8b30dd23e0..d6a7d3ea8d 100644 --- a/libavresample/dither.h +++ b/libavresample/dither.h @@ -66,7 +66,7 @@ typedef struct DitherDSPContext { DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, - int channels, int sample_rate); + int channels, int sample_rate, int apply_map); /** * Free a DitherContext. diff --git a/libavresample/internal.h b/libavresample/internal.h index c094f08f49..057f89a49c 100644 --- a/libavresample/internal.h +++ b/libavresample/internal.h @@ -32,6 +32,24 @@ typedef struct AudioConvert AudioConvert; typedef struct AudioMix AudioMix; typedef struct ResampleContext ResampleContext; +enum RemapPoint { + REMAP_NONE, + REMAP_IN_COPY, + REMAP_IN_CONVERT, + REMAP_OUT_COPY, + REMAP_OUT_CONVERT, +}; + +typedef struct ChannelMapInfo { + int channel_map[AVRESAMPLE_MAX_CHANNELS]; /**< source index of each output channel, -1 if not remapped */ + int do_remap; /**< remap needed */ + int channel_copy[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to copy from */ + int do_copy; /**< copy needed */ + int channel_zero[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to zero */ + int do_zero; /**< zeroing needed */ + int input_map[AVRESAMPLE_MAX_CHANNELS]; /**< dest index of each input channel */ +} ChannelMapInfo; + struct AVAudioResampleContext { const AVClass *av_class; /**< AVClass for logging and AVOptions */ @@ -65,6 +83,7 @@ struct AVAudioResampleContext { int resample_needed; /**< resampling is needed */ int in_convert_needed; /**< input sample format conversion is needed */ int out_convert_needed; /**< output sample format conversion is needed */ + int in_copy_needed; /**< input data copy is needed */ AudioData *in_buffer; /**< buffer for converted input */ AudioData *resample_out_buffer; /**< buffer for output from resampler */ @@ -82,6 +101,10 @@ struct AVAudioResampleContext { * only used if avresample_set_matrix() is called before avresample_open() */ double *mix_matrix; + + int use_channel_map; + enum RemapPoint remap_point; + ChannelMapInfo ch_map_info; }; #endif /* AVRESAMPLE_INTERNAL_H */ diff --git a/libavresample/utils.c b/libavresample/utils.c index a30388092e..b79def9285 100644 --- a/libavresample/utils.c +++ b/libavresample/utils.c @@ -96,20 +96,84 @@ int avresample_open(AVAudioResampleContext *avr) av_get_sample_fmt_name(avr->internal_sample_fmt)); } - /* set sample format conversion parameters */ + /* treat all mono as planar for easier comparison */ if (avr->in_channels == 1) avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt); if (avr->out_channels == 1) avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); - avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) && - avr->in_sample_fmt != avr->internal_sample_fmt; + + /* we may need to add an extra conversion in order to remap channels if + the output format is not planar */ + if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed && + !av_sample_fmt_is_planar(avr->out_sample_fmt)) { + avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); + } + + /* set sample format conversion parameters */ if (avr->resample_needed || avr->mixing_needed) + avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt; + else + avr->in_convert_needed = avr->use_channel_map && + !av_sample_fmt_is_planar(avr->out_sample_fmt); + + if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed) avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt; else avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt; + avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed || + (avr->use_channel_map && avr->resample_needed)); + + if (avr->use_channel_map) { + if (avr->in_copy_needed) { + avr->remap_point = REMAP_IN_COPY; + av_dlog(avr, "remap channels during in_copy\n"); + } else if (avr->in_convert_needed) { + avr->remap_point = REMAP_IN_CONVERT; + av_dlog(avr, "remap channels during in_convert\n"); + } else if (avr->out_convert_needed) { + avr->remap_point = REMAP_OUT_CONVERT; + av_dlog(avr, "remap channels during out_convert\n"); + } else { + avr->remap_point = REMAP_OUT_COPY; + av_dlog(avr, "remap channels during out_copy\n"); + } + +#ifdef DEBUG + { + int ch; + av_dlog(avr, "output map: "); + if (avr->ch_map_info.do_remap) + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]); + else + av_dlog(avr, "n/a"); + av_dlog(avr, "\n"); + av_dlog(avr, "copy map: "); + if (avr->ch_map_info.do_copy) + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]); + else + av_dlog(avr, "n/a"); + av_dlog(avr, "\n"); + av_dlog(avr, "zero map: "); + if (avr->ch_map_info.do_zero) + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]); + else + av_dlog(avr, "n/a"); + av_dlog(avr, "\n"); + av_dlog(avr, "input map: "); + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]); + av_dlog(avr, "\n"); + } +#endif + } else + avr->remap_point = REMAP_NONE; + /* allocate buffers */ - if (avr->mixing_needed || avr->in_convert_needed) { + if (avr->in_copy_needed || avr->in_convert_needed) { avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels), 0, avr->internal_sample_fmt, "in_buffer"); @@ -146,7 +210,8 @@ int avresample_open(AVAudioResampleContext *avr) if (avr->in_convert_needed) { avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt, avr->in_sample_fmt, avr->in_channels, - avr->in_sample_rate); + avr->in_sample_rate, + avr->remap_point == REMAP_IN_CONVERT); if (!avr->ac_in) { ret = AVERROR(ENOMEM); goto error; @@ -160,7 +225,8 @@ int avresample_open(AVAudioResampleContext *avr) src_fmt = avr->in_sample_fmt; avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt, avr->out_channels, - avr->out_sample_rate); + avr->out_sample_rate, + avr->remap_point == REMAP_OUT_CONVERT); if (!avr->ac_out) { ret = AVERROR(ENOMEM); goto error; @@ -200,6 +266,8 @@ void avresample_close(AVAudioResampleContext *avr) ff_audio_resample_free(&avr->resample); ff_audio_mix_free(&avr->am); av_freep(&avr->mix_matrix); + + avr->use_channel_map = 0; } void avresample_free(AVAudioResampleContext **avr) @@ -242,7 +310,9 @@ static int handle_buffered_output(AVAudioResampleContext *avr, data in the output FIFO */ av_dlog(avr, "[copy] %s to output\n", converted->name); output->nb_samples = 0; - ret = ff_audio_data_copy(output, converted); + ret = ff_audio_data_copy(output, converted, + avr->remap_point == REMAP_OUT_COPY ? + &avr->ch_map_info : NULL); if (ret < 0) return ret; av_dlog(avr, "[end conversion]\n"); @@ -306,11 +376,24 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr, /* in some rare cases we can copy input to output and upmix directly in the output buffer */ av_dlog(avr, "[copy] %s to output\n", current_buffer->name); - ret = ff_audio_data_copy(&output_buffer, current_buffer); + ret = ff_audio_data_copy(&output_buffer, current_buffer, + avr->remap_point == REMAP_OUT_COPY ? + &avr->ch_map_info : NULL); if (ret < 0) return ret; current_buffer = &output_buffer; - } else if (avr->mixing_needed || avr->in_convert_needed) { + } else if (avr->remap_point == REMAP_OUT_COPY && + (!direct_output || out_samples < in_samples)) { + /* if remapping channels during output copy, we may need to + * use an intermediate buffer in order to remap before adding + * samples to the output fifo */ + av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name); + ret = ff_audio_data_copy(avr->out_buffer, current_buffer, + &avr->ch_map_info); + if (ret < 0) + return ret; + current_buffer = avr->out_buffer; + } else if (avr->in_copy_needed || avr->in_convert_needed) { /* if needed, copy or convert input to in_buffer, and downmix if applicable */ if (avr->in_convert_needed) { @@ -325,7 +408,9 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr, return ret; } else { av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name); - ret = ff_audio_data_copy(avr->in_buffer, current_buffer); + ret = ff_audio_data_copy(avr->in_buffer, current_buffer, + avr->remap_point == REMAP_IN_COPY ? + &avr->ch_map_info : NULL); if (ret < 0) return ret; } @@ -470,6 +555,57 @@ int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, return 0; } +int avresample_set_channel_mapping(AVAudioResampleContext *avr, + const int *channel_map) +{ + ChannelMapInfo *info = &avr->ch_map_info; + int in_channels, ch, i; + + in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); + if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) { + av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n"); + return AVERROR(EINVAL); + } + + memset(info, 0, sizeof(*info)); + memset(info->input_map, -1, sizeof(info->input_map)); + + for (ch = 0; ch < in_channels; ch++) { + if (channel_map[ch] >= in_channels) { + av_log(avr, AV_LOG_ERROR, "Invalid channel map\n"); + return AVERROR(EINVAL); + } + if (channel_map[ch] < 0) { + info->channel_zero[ch] = 1; + info->channel_map[ch] = -1; + info->do_zero = 1; + } else if (info->input_map[channel_map[ch]] >= 0) { + info->channel_copy[ch] = info->input_map[channel_map[ch]]; + info->channel_map[ch] = -1; + info->do_copy = 1; + } else { + info->channel_map[ch] = channel_map[ch]; + info->input_map[channel_map[ch]] = ch; + info->do_remap = 1; + } + } + /* Fill-in unmapped input channels with unmapped output channels. + This is used when remapping during conversion from interleaved to + planar format. */ + for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) { + while (ch < in_channels && info->input_map[ch] >= 0) + ch++; + while (i < in_channels && info->channel_map[i] >= 0) + i++; + if (ch >= in_channels || i >= in_channels) + break; + info->input_map[ch] = i; + } + + avr->use_channel_map = 1; + return 0; +} + int avresample_available(AVAudioResampleContext *avr) { return av_audio_fifo_size(avr->out_fifo); diff --git a/libavresample/version.h b/libavresample/version.h index ebcd07f57c..387d097d3a 100644 --- a/libavresample/version.h +++ b/libavresample/version.h @@ -20,8 +20,8 @@ #define AVRESAMPLE_VERSION_H #define LIBAVRESAMPLE_VERSION_MAJOR 1 -#define LIBAVRESAMPLE_VERSION_MINOR 0 -#define LIBAVRESAMPLE_VERSION_MICRO 1 +#define LIBAVRESAMPLE_VERSION_MINOR 1 +#define LIBAVRESAMPLE_VERSION_MICRO 0 #define LIBAVRESAMPLE_VERSION_INT AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \ LIBAVRESAMPLE_VERSION_MINOR, \ -- cgit v1.2.3