diff --git a/avconv.c b/avconv.c index 6c3a01f6b6..5bd5121151 100644 --- a/avconv.c +++ b/avconv.c @@ -31,8 +31,8 @@ #include "libavformat/avformat.h" #include "libavdevice/avdevice.h" #include "libswscale/swscale.h" +#include "libavresample/avresample.h" #include "libavutil/opt.h" -#include "libavcodec/audioconvert.h" #include "libavutil/audioconvert.h" #include "libavutil/parseutils.h" #include "libavutil/samplefmt.h" @@ -266,12 +266,11 @@ typedef struct OutputStream { /* audio only */ int audio_resample; - ReSampleContext *resample; /* for audio resampling */ + AVAudioResampleContext *avr; int resample_sample_fmt; int resample_channels; + uint64_t resample_channel_layout; int resample_sample_rate; - int reformat_pair; - AVAudioConvert *reformat_ctx; AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */ FILE *logfile; @@ -1314,7 +1313,7 @@ static int encode_audio_frame(AVFormatContext *s, OutputStream *ost, } static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc, - int nb_samples) + int nb_samples, int *buf_linesize) { int64_t audio_buf_samples; int audio_buf_size; @@ -1327,7 +1326,7 @@ static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc, if (audio_buf_samples > INT_MAX) return AVERROR(EINVAL); - audio_buf_size = av_samples_get_buffer_size(NULL, enc->channels, + audio_buf_size = av_samples_get_buffer_size(buf_linesize, enc->channels, audio_buf_samples, enc->sample_fmt, 0); if (audio_buf_size < 0) @@ -1345,77 +1344,88 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost, { uint8_t *buftmp; - int size_out, frame_bytes, resample_changed; + int size_out, frame_bytes, resample_changed, ret; AVCodecContext *enc = ost->st->codec; AVCodecContext *dec = ist->st->codec; int osize = av_get_bytes_per_sample(enc->sample_fmt); int isize = av_get_bytes_per_sample(dec->sample_fmt); uint8_t *buf = decoded_frame->data[0]; int size = decoded_frame->nb_samples * dec->channels * isize; + int out_linesize = 0; + int buf_linesize = decoded_frame->linesize[0]; get_default_channel_layouts(ost, ist); - if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples) < 0) { + if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples, &out_linesize) < 0) { av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n"); exit_program(1); } - if (enc->channels != dec->channels || enc->sample_rate != dec->sample_rate) + if (audio_sync_method > 1 || + enc->channels != dec->channels || + enc->channel_layout != dec->channel_layout || + enc->sample_rate != dec->sample_rate || + dec->sample_fmt != enc->sample_fmt) ost->audio_resample = 1; resample_changed = ost->resample_sample_fmt != dec->sample_fmt || ost->resample_channels != dec->channels || + ost->resample_channel_layout != dec->channel_layout || ost->resample_sample_rate != dec->sample_rate; - if ((ost->audio_resample && !ost->resample) || resample_changed) { + if ((ost->audio_resample && !ost->avr) || resample_changed) { if (resample_changed) { - av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n", + av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d chl:0x%"PRIx64" to rate:%d fmt:%s ch:%d chl:0x%"PRIx64"\n", ist->file_index, ist->st->index, - ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), ost->resample_channels, - dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels); + ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), + ost->resample_channels, ost->resample_channel_layout, + dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), + dec->channels, dec->channel_layout); ost->resample_sample_fmt = dec->sample_fmt; ost->resample_channels = dec->channels; + ost->resample_channel_layout = dec->channel_layout; ost->resample_sample_rate = dec->sample_rate; - if (ost->resample) - audio_resample_close(ost->resample); + if (ost->avr) + avresample_close(ost->avr); } /* if audio_sync_method is >1 the resampler is needed for audio drift compensation */ if (audio_sync_method <= 1 && ost->resample_sample_fmt == enc->sample_fmt && ost->resample_channels == enc->channels && + ost->resample_channel_layout == enc->channel_layout && ost->resample_sample_rate == enc->sample_rate) { - ost->resample = NULL; ost->audio_resample = 0; } else if (ost->audio_resample) { - if (dec->sample_fmt != AV_SAMPLE_FMT_S16) - av_log(NULL, AV_LOG_WARNING, "Using s16 intermediate sample format for resampling\n"); - ost->resample = av_audio_resample_init(enc->channels, dec->channels, - enc->sample_rate, dec->sample_rate, - enc->sample_fmt, dec->sample_fmt, - 16, 10, 0, 0.8); - if (!ost->resample) { - av_log(NULL, AV_LOG_FATAL, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n", - dec->channels, dec->sample_rate, - enc->channels, enc->sample_rate); - exit_program(1); + if (!ost->avr) { + ost->avr = avresample_alloc_context(); + if (!ost->avr) { + av_log(NULL, AV_LOG_FATAL, "Error allocating context for libavresample\n"); + exit_program(1); + } } - } - } -#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b)) - if (!ost->audio_resample && dec->sample_fmt != enc->sample_fmt && - MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt) != ost->reformat_pair) { - if (ost->reformat_ctx) - av_audio_convert_free(ost->reformat_ctx); - ost->reformat_ctx = av_audio_convert_alloc(enc->sample_fmt, 1, - dec->sample_fmt, 1, NULL, 0); - if (!ost->reformat_ctx) { - av_log(NULL, AV_LOG_FATAL, "Cannot convert %s sample format to %s sample format\n", - av_get_sample_fmt_name(dec->sample_fmt), - av_get_sample_fmt_name(enc->sample_fmt)); - exit_program(1); + av_opt_set_int(ost->avr, "in_channel_layout", dec->channel_layout, 0); + av_opt_set_int(ost->avr, "in_sample_fmt", dec->sample_fmt, 0); + av_opt_set_int(ost->avr, "in_sample_rate", dec->sample_rate, 0); + av_opt_set_int(ost->avr, "out_channel_layout", enc->channel_layout, 0); + av_opt_set_int(ost->avr, "out_sample_fmt", enc->sample_fmt, 0); + av_opt_set_int(ost->avr, "out_sample_rate", enc->sample_rate, 0); + if (audio_sync_method > 1) + av_opt_set_int(ost->avr, "force_resampling", 1, 0); + + /* if both the input and output formats are s16 or u8, use s16 as + the internal sample format */ + if (av_get_bytes_per_sample(dec->sample_fmt) <= 2 && + av_get_bytes_per_sample(enc->sample_fmt) <= 2) { + av_opt_set_int(ost->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0); + } + + ret = avresample_open(ost->avr); + if (ret < 0) { + av_log(NULL, AV_LOG_FATAL, "Error opening libavresample\n"); + exit_program(1); + } } - ost->reformat_pair = MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt); } if (audio_sync_method > 0) { @@ -1444,7 +1454,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost, exit_program(1); } - if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta) < 0) { + if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta, &out_linesize) < 0) { av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n"); exit_program(1); } @@ -1454,15 +1464,15 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost, memcpy(async_buf + byte_delta, buf, size); buf = async_buf; size += byte_delta; + buf_linesize = allocated_async_buf_size; av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", idelta); } } else if (audio_sync_method > 1) { int comp = av_clip(delta, -audio_sync_method, audio_sync_method); - av_assert0(ost->audio_resample); av_log(NULL, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate); // fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(ost->fifo)/(ost->st->codec->channels * 2)); - av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate); + avresample_set_compensation(ost->avr, comp, enc->sample_rate); } } } else if (audio_sync_method == 0) @@ -1471,31 +1481,16 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost, if (ost->audio_resample) { buftmp = audio_buf; - size_out = audio_resample(ost->resample, - (short *)buftmp, (short *)buf, - size / (dec->channels * isize)); + size_out = avresample_convert(ost->avr, (void **)&buftmp, + allocated_audio_buf_size, out_linesize, + (void **)&buf, buf_linesize, + size / (dec->channels * isize)); size_out = size_out * enc->channels * osize; } else { buftmp = buf; size_out = size; } - if (!ost->audio_resample && dec->sample_fmt != enc->sample_fmt) { - const void *ibuf[6] = { buftmp }; - void *obuf[6] = { audio_buf }; - int istride[6] = { isize }; - int ostride[6] = { osize }; - int len = size_out / istride[0]; - if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len) < 0) { - printf("av_audio_convert() failed\n"); - if (exit_on_error) - exit_program(1); - return; - } - buftmp = audio_buf; - size_out = len * osize; - } - /* now encode as many frames as possible */ if (!(enc->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) { /* output resampled raw samples */ @@ -2709,7 +2704,6 @@ static int transcode_init(void) if (!ost->fifo) { return AVERROR(ENOMEM); } - ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE); if (!codec->sample_rate) codec->sample_rate = icodec->sample_rate; @@ -2722,15 +2716,16 @@ static int transcode_init(void) if (!codec->channels) codec->channels = icodec->channels; - codec->channel_layout = icodec->channel_layout; + if (!codec->channel_layout) + codec->channel_layout = icodec->channel_layout; if (av_get_channel_layout_nb_channels(codec->channel_layout) != codec->channels) codec->channel_layout = 0; - ost->audio_resample = codec-> sample_rate != icodec->sample_rate || audio_sync_method > 1; icodec->request_channels = codec-> channels; ost->resample_sample_fmt = icodec->sample_fmt; ost->resample_sample_rate = icodec->sample_rate; ost->resample_channels = icodec->channels; + ost->resample_channel_layout = icodec->channel_layout; break; case AVMEDIA_TYPE_VIDEO: if (!ost->filter) { @@ -3202,10 +3197,8 @@ static int transcode(void) initialized but set to zero */ av_freep(&ost->st->codec->subtitle_header); av_free(ost->forced_kf_pts); - if (ost->resample) - audio_resample_close(ost->resample); - if (ost->reformat_ctx) - av_audio_convert_free(ost->reformat_ctx); + if (ost->avr) + avresample_free(&ost->avr); av_dict_free(&ost->opts); } } diff --git a/cmdutils.c b/cmdutils.c index d590d0a49f..6d2e97f694 100644 --- a/cmdutils.c +++ b/cmdutils.c @@ -32,6 +32,7 @@ #include "libavformat/avformat.h" #include "libavfilter/avfilter.h" #include "libavdevice/avdevice.h" +#include "libavresample/avresample.h" #include "libswscale/swscale.h" #include "libavutil/avstring.h" #include "libavutil/mathematics.h" @@ -460,7 +461,8 @@ static int warned_cfg = 0; const char *indent = flags & INDENT? " " : ""; \ if (flags & SHOW_VERSION) { \ unsigned int version = libname##_version(); \ - av_log(NULL, level, "%slib%-9s %2d.%3d.%2d / %2d.%3d.%2d\n",\ + av_log(NULL, level, \ + "%slib%-10s %2d.%3d.%2d / %2d.%3d.%2d\n", \ indent, #libname, \ LIB##LIBNAME##_VERSION_MAJOR, \ LIB##LIBNAME##_VERSION_MINOR, \ @@ -489,6 +491,7 @@ static void print_all_libs_info(int flags, int level) PRINT_LIB_INFO(avformat, AVFORMAT, flags, level); PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level); PRINT_LIB_INFO(avfilter, AVFILTER, flags, level); + PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level); PRINT_LIB_INFO(swscale, SWSCALE, flags, level); }