FFmpeg: libavresample/utils.c Source File

00001 /*
00002  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
00003  *
00004  * This file is part of Libav.
00005  *
00006  * Libav is free software; you can redistribute it and/or
00007  * modify it under the terms of the GNU Lesser General Public
00008  * License as published by the Free Software Foundation; either
00009  * version 2.1 of the License, or (at your option) any later version.
00010  *
00011  * Libav is distributed in the hope that it will be useful,
00012  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00013  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00014  * Lesser General Public License for more details.
00015  *
00016  * You should have received a copy of the GNU Lesser General Public
00017  * License along with Libav; if not, write to the Free Software
00018  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00019  */
00020 
00021 #include "libavutil/common.h"
00022 #include "libavutil/dict.h"
00023 // #include "libavutil/error.h"
00024 #include "libavutil/log.h"
00025 #include "libavutil/mem.h"
00026 #include "libavutil/opt.h"
00027 
00028 #include "avresample.h"
00029 #include "audio_data.h"
00030 #include "internal.h"
00031 
00032 int avresample_open(AVAudioResampleContext *avr)
00033 {
00034     int ret;
00035 
00036     /* set channel mixing parameters */
00037     avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
00038     if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
00039         av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
00040                avr->in_channel_layout);
00041         return AVERROR(EINVAL);
00042     }
00043     avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
00044     if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
00045         av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
00046                avr->out_channel_layout);
00047         return AVERROR(EINVAL);
00048     }
00049     avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
00050     avr->downmix_needed    = avr->in_channels  > avr->out_channels;
00051     avr->upmix_needed      = avr->out_channels > avr->in_channels ||
00052                              (!avr->downmix_needed && (avr->am->matrix ||
00053                               avr->in_channel_layout != avr->out_channel_layout));
00054     avr->mixing_needed     = avr->downmix_needed || avr->upmix_needed;
00055 
00056     /* set resampling parameters */
00057     avr->resample_needed   = avr->in_sample_rate != avr->out_sample_rate ||
00058                              avr->force_resampling;
00059 
00060     /* select internal sample format if not specified by the user */
00061     if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
00062         (avr->mixing_needed || avr->resample_needed)) {
00063         enum AVSampleFormat  in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
00064         enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
00065         int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
00066                             av_get_bytes_per_sample(out_fmt));
00067         if (max_bps <= 2) {
00068             avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
00069         } else if (avr->mixing_needed) {
00070             avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
00071         } else {
00072             if (max_bps <= 4) {
00073                 if (in_fmt  == AV_SAMPLE_FMT_S32P ||
00074                     out_fmt == AV_SAMPLE_FMT_S32P) {
00075                     if (in_fmt  == AV_SAMPLE_FMT_FLTP ||
00076                         out_fmt == AV_SAMPLE_FMT_FLTP) {
00077                         /* if one is s32 and the other is flt, use dbl */
00078                         avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
00079                     } else {
00080                         /* if one is s32 and the other is s32, s16, or u8, use s32 */
00081                         avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
00082                     }
00083                 } else {
00084                     /* if one is flt and the other is flt, s16 or u8, use flt */
00085                     avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
00086                 }
00087             } else {
00088                 /* if either is dbl, use dbl */
00089                 avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
00090             }
00091         }
00092         av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
00093                av_get_sample_fmt_name(avr->internal_sample_fmt));
00094     }
00095 
00096     /* set sample format conversion parameters */
00097     if (avr->in_channels == 1)
00098         avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
00099     if (avr->out_channels == 1)
00100         avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
00101     avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
00102                               avr->in_sample_fmt != avr->internal_sample_fmt;
00103     if (avr->resample_needed || avr->mixing_needed)
00104         avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
00105     else
00106         avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
00107 
00108     /* allocate buffers */
00109     if (avr->mixing_needed || avr->in_convert_needed) {
00110         avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
00111                                              0, avr->internal_sample_fmt,
00112                                              "in_buffer");
00113         if (!avr->in_buffer) {
00114             ret = AVERROR(EINVAL);
00115             goto error;
00116         }
00117     }
00118     if (avr->resample_needed) {
00119         avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
00120                                                        0, avr->internal_sample_fmt,
00121                                                        "resample_out_buffer");
00122         if (!avr->resample_out_buffer) {
00123             ret = AVERROR(EINVAL);
00124             goto error;
00125         }
00126     }
00127     if (avr->out_convert_needed) {
00128         avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
00129                                               avr->out_sample_fmt, "out_buffer");
00130         if (!avr->out_buffer) {
00131             ret = AVERROR(EINVAL);
00132             goto error;
00133         }
00134     }
00135     avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
00136                                         1024);
00137     if (!avr->out_fifo) {
00138         ret = AVERROR(ENOMEM);
00139         goto error;
00140     }
00141 
00142     /* setup contexts */
00143     if (avr->in_convert_needed) {
00144         avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
00145                                             avr->in_sample_fmt, avr->in_channels);
00146         if (!avr->ac_in) {
00147             ret = AVERROR(ENOMEM);
00148             goto error;
00149         }
00150     }
00151     if (avr->out_convert_needed) {
00152         enum AVSampleFormat src_fmt;
00153         if (avr->in_convert_needed)
00154             src_fmt = avr->internal_sample_fmt;
00155         else
00156             src_fmt = avr->in_sample_fmt;
00157         avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
00158                                              avr->out_channels);
00159         if (!avr->ac_out) {
00160             ret = AVERROR(ENOMEM);
00161             goto error;
00162         }
00163     }
00164     if (avr->resample_needed) {
00165         avr->resample = ff_audio_resample_init(avr);
00166         if (!avr->resample) {
00167             ret = AVERROR(ENOMEM);
00168             goto error;
00169         }
00170     }
00171     if (avr->mixing_needed) {
00172         ret = ff_audio_mix_init(avr);
00173         if (ret < 0)
00174             goto error;
00175     }
00176 
00177     return 0;
00178 
00179 error:
00180     avresample_close(avr);
00181     return ret;
00182 }
00183 
00184 void avresample_close(AVAudioResampleContext *avr)
00185 {
00186     ff_audio_data_free(&avr->in_buffer);
00187     ff_audio_data_free(&avr->resample_out_buffer);
00188     ff_audio_data_free(&avr->out_buffer);
00189     av_audio_fifo_free(avr->out_fifo);
00190     avr->out_fifo = NULL;
00191     av_freep(&avr->ac_in);
00192     av_freep(&avr->ac_out);
00193     ff_audio_resample_free(&avr->resample);
00194     ff_audio_mix_close(avr->am);
00195     return;
00196 }
00197 
00198 void avresample_free(AVAudioResampleContext **avr)
00199 {
00200     if (!*avr)
00201         return;
00202     avresample_close(*avr);
00203     av_freep(&(*avr)->am);
00204     av_opt_free(*avr);
00205     av_freep(avr);
00206 }
00207 
00208 static int handle_buffered_output(AVAudioResampleContext *avr,
00209                                   AudioData *output, AudioData *converted)
00210 {
00211     int ret;
00212 
00213     if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
00214         (converted && output->allocated_samples < converted->nb_samples)) {
00215         if (converted) {
00216             /* if there are any samples in the output FIFO or if the
00217                user-supplied output buffer is not large enough for all samples,
00218                we add to the output FIFO */
00219             av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name);
00220             ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
00221                                             converted->nb_samples);
00222             if (ret < 0)
00223                 return ret;
00224         }
00225 
00226         /* if the user specified an output buffer, read samples from the output
00227            FIFO to the user output */
00228         if (output && output->allocated_samples > 0) {
00229             av_dlog(avr, "[FIFO] read from out_fifo to output\n");
00230             av_dlog(avr, "[end conversion]\n");
00231             return ff_audio_data_read_from_fifo(avr->out_fifo, output,
00232                                                 output->allocated_samples);
00233         }
00234     } else if (converted) {
00235         /* copy directly to output if it is large enough or there is not any
00236            data in the output FIFO */
00237         av_dlog(avr, "[copy] %s to output\n", converted->name);
00238         output->nb_samples = 0;
00239         ret = ff_audio_data_copy(output, converted);
00240         if (ret < 0)
00241             return ret;
00242         av_dlog(avr, "[end conversion]\n");
00243         return output->nb_samples;
00244     }
00245     av_dlog(avr, "[end conversion]\n");
00246     return 0;
00247 }
00248 
00249 int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
00250                                            void **output, int out_plane_size,
00251                                            int out_samples, void **input,
00252                                            int in_plane_size, int in_samples)
00253 {
00254     AudioData input_buffer;
00255     AudioData output_buffer;
00256     AudioData *current_buffer;
00257     int ret;
00258 
00259     /* reset internal buffers */
00260     if (avr->in_buffer) {
00261         avr->in_buffer->nb_samples = 0;
00262         ff_audio_data_set_channels(avr->in_buffer,
00263                                    avr->in_buffer->allocated_channels);
00264     }
00265     if (avr->resample_out_buffer) {
00266         avr->resample_out_buffer->nb_samples = 0;
00267         ff_audio_data_set_channels(avr->resample_out_buffer,
00268                                    avr->resample_out_buffer->allocated_channels);
00269     }
00270     if (avr->out_buffer) {
00271         avr->out_buffer->nb_samples = 0;
00272         ff_audio_data_set_channels(avr->out_buffer,
00273                                    avr->out_buffer->allocated_channels);
00274     }
00275 
00276     av_dlog(avr, "[start conversion]\n");
00277 
00278     /* initialize output_buffer with output data */
00279     if (output) {
00280         ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
00281                                  avr->out_channels, out_samples,
00282                                  avr->out_sample_fmt, 0, "output");
00283         if (ret < 0)
00284             return ret;
00285         output_buffer.nb_samples = 0;
00286     }
00287 
00288     if (input) {
00289         /* initialize input_buffer with input data */
00290         ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
00291                                  avr->in_channels, in_samples,
00292                                  avr->in_sample_fmt, 1, "input");
00293         if (ret < 0)
00294             return ret;
00295         current_buffer = &input_buffer;
00296 
00297         if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
00298             !avr->out_convert_needed && output && out_samples >= in_samples) {
00299             /* in some rare cases we can copy input to output and upmix
00300                directly in the output buffer */
00301             av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
00302             ret = ff_audio_data_copy(&output_buffer, current_buffer);
00303             if (ret < 0)
00304                 return ret;
00305             current_buffer = &output_buffer;
00306         } else if (avr->mixing_needed || avr->in_convert_needed) {
00307             /* if needed, copy or convert input to in_buffer, and downmix if
00308                applicable */
00309             if (avr->in_convert_needed) {
00310                 ret = ff_audio_data_realloc(avr->in_buffer,
00311                                             current_buffer->nb_samples);
00312                 if (ret < 0)
00313                     return ret;
00314                 av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
00315                 ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer,
00316                                        current_buffer->nb_samples);
00317                 if (ret < 0)
00318                     return ret;
00319             } else {
00320                 av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
00321                 ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
00322                 if (ret < 0)
00323                     return ret;
00324             }
00325             ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
00326             if (avr->downmix_needed) {
00327                 av_dlog(avr, "[downmix] in_buffer\n");
00328                 ret = ff_audio_mix(avr->am, avr->in_buffer);
00329                 if (ret < 0)
00330                     return ret;
00331             }
00332             current_buffer = avr->in_buffer;
00333         }
00334     } else {
00335         /* flush resampling buffer and/or output FIFO if input is NULL */
00336         if (!avr->resample_needed)
00337             return handle_buffered_output(avr, output ? &output_buffer : NULL,
00338                                           NULL);
00339         current_buffer = NULL;
00340     }
00341 
00342     if (avr->resample_needed) {
00343         AudioData *resample_out;
00344         int consumed = 0;
00345 
00346         if (!avr->out_convert_needed && output && out_samples > 0)
00347             resample_out = &output_buffer;
00348         else
00349             resample_out = avr->resample_out_buffer;
00350         av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
00351                 resample_out->name);
00352         ret = ff_audio_resample(avr->resample, resample_out,
00353                                 current_buffer, &consumed);
00354         if (ret < 0)
00355             return ret;
00356 
00357         /* if resampling did not produce any samples, just return 0 */
00358         if (resample_out->nb_samples == 0) {
00359             av_dlog(avr, "[end conversion]\n");
00360             return 0;
00361         }
00362 
00363         current_buffer = resample_out;
00364     }
00365 
00366     if (avr->upmix_needed) {
00367         av_dlog(avr, "[upmix] %s\n", current_buffer->name);
00368         ret = ff_audio_mix(avr->am, current_buffer);
00369         if (ret < 0)
00370             return ret;
00371     }
00372 
00373     /* if we resampled or upmixed directly to output, return here */
00374     if (current_buffer == &output_buffer) {
00375         av_dlog(avr, "[end conversion]\n");
00376         return current_buffer->nb_samples;
00377     }
00378 
00379     if (avr->out_convert_needed) {
00380         if (output && out_samples >= current_buffer->nb_samples) {
00381             /* convert directly to output */
00382             av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
00383             ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer,
00384                                    current_buffer->nb_samples);
00385             if (ret < 0)
00386                 return ret;
00387 
00388             av_dlog(avr, "[end conversion]\n");
00389             return output_buffer.nb_samples;
00390         } else {
00391             ret = ff_audio_data_realloc(avr->out_buffer,
00392                                         current_buffer->nb_samples);
00393             if (ret < 0)
00394                 return ret;
00395             av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
00396             ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
00397                                    current_buffer, current_buffer->nb_samples);
00398             if (ret < 0)
00399                 return ret;
00400             current_buffer = avr->out_buffer;
00401         }
00402     }
00403 
00404     return handle_buffered_output(avr, output ? &output_buffer : NULL,
00405                                   current_buffer);
00406 }
00407 
00408 int avresample_available(AVAudioResampleContext *avr)
00409 {
00410     return av_audio_fifo_size(avr->out_fifo);
00411 }
00412 
00413 int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples)
00414 {
00415     if (!output)
00416         return av_audio_fifo_drain(avr->out_fifo, nb_samples);
00417     return av_audio_fifo_read(avr->out_fifo, output, nb_samples);
00418 }
00419 
00420 unsigned avresample_version(void)
00421 {
00422     return LIBAVRESAMPLE_VERSION_INT;
00423 }
00424 
00425 const char *avresample_license(void)
00426 {
00427 #define LICENSE_PREFIX "libavresample license: "
00428     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
00429 }
00430 
00431 const char *avresample_configuration(void)
00432 {
00433     return FFMPEG_CONFIGURATION;
00434 }