| /* |
| * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
| * |
| * This file is part of Libav. |
| * |
| * Libav is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser General Public |
| * License as published by the Free Software Foundation; either |
| * version 2.1 of the License, or (at your option) any later version. |
| * |
| * Libav is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with Libav; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| |
| #include "libavutil/dict.h" |
| // #include "libavutil/error.h" |
| #include "libavutil/log.h" |
| #include "libavutil/mem.h" |
| #include "libavutil/opt.h" |
| |
| #include "avresample.h" |
| #include "audio_data.h" |
| #include "internal.h" |
| |
| int avresample_open(AVAudioResampleContext *avr) |
| { |
| int ret; |
| |
| /* set channel mixing parameters */ |
| avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); |
| if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) { |
| av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n", |
| avr->in_channel_layout); |
| return AVERROR(EINVAL); |
| } |
| avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); |
| if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) { |
| av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n", |
| avr->out_channel_layout); |
| return AVERROR(EINVAL); |
| } |
| avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels); |
| avr->downmix_needed = avr->in_channels > avr->out_channels; |
| avr->upmix_needed = avr->out_channels > avr->in_channels || |
| avr->am->matrix || |
| (avr->out_channels == avr->in_channels && |
| avr->in_channel_layout != avr->out_channel_layout); |
| avr->mixing_needed = avr->downmix_needed || avr->upmix_needed; |
| |
| /* set resampling parameters */ |
| avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate || |
| avr->force_resampling; |
| |
| /* set sample format conversion parameters */ |
| /* override user-requested internal format to avoid unexpected failures |
| TODO: support more internal formats */ |
| if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) { |
| av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n"); |
| avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P; |
| } else if (avr->mixing_needed && |
| avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P && |
| avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) { |
| av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n"); |
| avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP; |
| } |
| 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; |
| if (avr->resample_needed || avr->mixing_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; |
| |
| /* allocate buffers */ |
| if (avr->mixing_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"); |
| if (!avr->in_buffer) { |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| } |
| if (avr->resample_needed) { |
| avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels, |
| 0, avr->internal_sample_fmt, |
| "resample_out_buffer"); |
| if (!avr->resample_out_buffer) { |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| } |
| if (avr->out_convert_needed) { |
| avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0, |
| avr->out_sample_fmt, "out_buffer"); |
| if (!avr->out_buffer) { |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| } |
| avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels, |
| 1024); |
| if (!avr->out_fifo) { |
| ret = AVERROR(ENOMEM); |
| goto error; |
| } |
| |
| /* setup contexts */ |
| if (avr->in_convert_needed) { |
| avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt, |
| avr->in_sample_fmt, avr->in_channels); |
| if (!avr->ac_in) { |
| ret = AVERROR(ENOMEM); |
| goto error; |
| } |
| } |
| if (avr->out_convert_needed) { |
| enum AVSampleFormat src_fmt; |
| if (avr->in_convert_needed) |
| src_fmt = avr->internal_sample_fmt; |
| else |
| src_fmt = avr->in_sample_fmt; |
| avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt, |
| avr->out_channels); |
| if (!avr->ac_out) { |
| ret = AVERROR(ENOMEM); |
| goto error; |
| } |
| } |
| if (avr->resample_needed) { |
| avr->resample = ff_audio_resample_init(avr); |
| if (!avr->resample) { |
| ret = AVERROR(ENOMEM); |
| goto error; |
| } |
| } |
| if (avr->mixing_needed) { |
| ret = ff_audio_mix_init(avr); |
| if (ret < 0) |
| goto error; |
| } |
| |
| return 0; |
| |
| error: |
| avresample_close(avr); |
| return ret; |
| } |
| |
| void avresample_close(AVAudioResampleContext *avr) |
| { |
| ff_audio_data_free(&avr->in_buffer); |
| ff_audio_data_free(&avr->resample_out_buffer); |
| ff_audio_data_free(&avr->out_buffer); |
| av_audio_fifo_free(avr->out_fifo); |
| avr->out_fifo = NULL; |
| av_freep(&avr->ac_in); |
| av_freep(&avr->ac_out); |
| ff_audio_resample_free(&avr->resample); |
| ff_audio_mix_close(avr->am); |
| return; |
| } |
| |
| void avresample_free(AVAudioResampleContext **avr) |
| { |
| if (!*avr) |
| return; |
| avresample_close(*avr); |
| av_freep(&(*avr)->am); |
| av_opt_free(*avr); |
| av_freep(avr); |
| } |
| |
| static int handle_buffered_output(AVAudioResampleContext *avr, |
| AudioData *output, AudioData *converted) |
| { |
| int ret; |
| |
| if (!output || av_audio_fifo_size(avr->out_fifo) > 0 || |
| (converted && output->allocated_samples < converted->nb_samples)) { |
| if (converted) { |
| /* if there are any samples in the output FIFO or if the |
| user-supplied output buffer is not large enough for all samples, |
| we add to the output FIFO */ |
| av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name); |
| ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0, |
| converted->nb_samples); |
| if (ret < 0) |
| return ret; |
| } |
| |
| /* if the user specified an output buffer, read samples from the output |
| FIFO to the user output */ |
| if (output && output->allocated_samples > 0) { |
| av_dlog(avr, "[FIFO] read from out_fifo to output\n"); |
| av_dlog(avr, "[end conversion]\n"); |
| return ff_audio_data_read_from_fifo(avr->out_fifo, output, |
| output->allocated_samples); |
| } |
| } else if (converted) { |
| /* copy directly to output if it is large enough or there is not any |
| 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); |
| if (ret < 0) |
| return ret; |
| av_dlog(avr, "[end conversion]\n"); |
| return output->nb_samples; |
| } |
| av_dlog(avr, "[end conversion]\n"); |
| return 0; |
| } |
| |
| int avresample_convert(AVAudioResampleContext *avr, void **output, |
| int out_plane_size, int out_samples, void **input, |
| int in_plane_size, int in_samples) |
| { |
| AudioData input_buffer; |
| AudioData output_buffer; |
| AudioData *current_buffer; |
| int ret; |
| |
| /* reset internal buffers */ |
| if (avr->in_buffer) { |
| avr->in_buffer->nb_samples = 0; |
| ff_audio_data_set_channels(avr->in_buffer, |
| avr->in_buffer->allocated_channels); |
| } |
| if (avr->resample_out_buffer) { |
| avr->resample_out_buffer->nb_samples = 0; |
| ff_audio_data_set_channels(avr->resample_out_buffer, |
| avr->resample_out_buffer->allocated_channels); |
| } |
| if (avr->out_buffer) { |
| avr->out_buffer->nb_samples = 0; |
| ff_audio_data_set_channels(avr->out_buffer, |
| avr->out_buffer->allocated_channels); |
| } |
| |
| av_dlog(avr, "[start conversion]\n"); |
| |
| /* initialize output_buffer with output data */ |
| if (output) { |
| ret = ff_audio_data_init(&output_buffer, output, out_plane_size, |
| avr->out_channels, out_samples, |
| avr->out_sample_fmt, 0, "output"); |
| if (ret < 0) |
| return ret; |
| output_buffer.nb_samples = 0; |
| } |
| |
| if (input) { |
| /* initialize input_buffer with input data */ |
| ret = ff_audio_data_init(&input_buffer, input, in_plane_size, |
| avr->in_channels, in_samples, |
| avr->in_sample_fmt, 1, "input"); |
| if (ret < 0) |
| return ret; |
| current_buffer = &input_buffer; |
| |
| if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed && |
| !avr->out_convert_needed && output && out_samples >= in_samples) { |
| /* 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); |
| if (ret < 0) |
| return ret; |
| current_buffer = &output_buffer; |
| } else if (avr->mixing_needed || avr->in_convert_needed) { |
| /* if needed, copy or convert input to in_buffer, and downmix if |
| applicable */ |
| if (avr->in_convert_needed) { |
| ret = ff_audio_data_realloc(avr->in_buffer, |
| current_buffer->nb_samples); |
| if (ret < 0) |
| return ret; |
| av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name); |
| ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer, |
| current_buffer->nb_samples); |
| if (ret < 0) |
| 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); |
| if (ret < 0) |
| return ret; |
| } |
| ff_audio_data_set_channels(avr->in_buffer, avr->in_channels); |
| if (avr->downmix_needed) { |
| av_dlog(avr, "[downmix] in_buffer\n"); |
| ret = ff_audio_mix(avr->am, avr->in_buffer); |
| if (ret < 0) |
| return ret; |
| } |
| current_buffer = avr->in_buffer; |
| } |
| } else { |
| /* flush resampling buffer and/or output FIFO if input is NULL */ |
| if (!avr->resample_needed) |
| return handle_buffered_output(avr, output ? &output_buffer : NULL, |
| NULL); |
| current_buffer = NULL; |
| } |
| |
| if (avr->resample_needed) { |
| AudioData *resample_out; |
| int consumed = 0; |
| |
| if (!avr->out_convert_needed && output && out_samples > 0) |
| resample_out = &output_buffer; |
| else |
| resample_out = avr->resample_out_buffer; |
| av_dlog(avr, "[resample] %s to %s\n", current_buffer->name, |
| resample_out->name); |
| ret = ff_audio_resample(avr->resample, resample_out, |
| current_buffer, &consumed); |
| if (ret < 0) |
| return ret; |
| |
| /* if resampling did not produce any samples, just return 0 */ |
| if (resample_out->nb_samples == 0) { |
| av_dlog(avr, "[end conversion]\n"); |
| return 0; |
| } |
| |
| current_buffer = resample_out; |
| } |
| |
| if (avr->upmix_needed) { |
| av_dlog(avr, "[upmix] %s\n", current_buffer->name); |
| ret = ff_audio_mix(avr->am, current_buffer); |
| if (ret < 0) |
| return ret; |
| } |
| |
| /* if we resampled or upmixed directly to output, return here */ |
| if (current_buffer == &output_buffer) { |
| av_dlog(avr, "[end conversion]\n"); |
| return current_buffer->nb_samples; |
| } |
| |
| if (avr->out_convert_needed) { |
| if (output && out_samples >= current_buffer->nb_samples) { |
| /* convert directly to output */ |
| av_dlog(avr, "[convert] %s to output\n", current_buffer->name); |
| ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer, |
| current_buffer->nb_samples); |
| if (ret < 0) |
| return ret; |
| |
| av_dlog(avr, "[end conversion]\n"); |
| return output_buffer.nb_samples; |
| } else { |
| ret = ff_audio_data_realloc(avr->out_buffer, |
| current_buffer->nb_samples); |
| if (ret < 0) |
| return ret; |
| av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name); |
| ret = ff_audio_convert(avr->ac_out, avr->out_buffer, |
| current_buffer, current_buffer->nb_samples); |
| if (ret < 0) |
| return ret; |
| current_buffer = avr->out_buffer; |
| } |
| } |
| |
| return handle_buffered_output(avr, output ? &output_buffer : NULL, |
| current_buffer); |
| } |
| |
| int avresample_available(AVAudioResampleContext *avr) |
| { |
| return av_audio_fifo_size(avr->out_fifo); |
| } |
| |
| int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples) |
| { |
| if (!output) |
| return av_audio_fifo_drain(avr->out_fifo, nb_samples); |
| return av_audio_fifo_read(avr->out_fifo, output, nb_samples); |
| } |
| |
| unsigned avresample_version(void) |
| { |
| return LIBAVRESAMPLE_VERSION_INT; |
| } |
| |
| const char *avresample_license(void) |
| { |
| #define LICENSE_PREFIX "libavresample license: " |
| return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1; |
| } |
| |
| const char *avresample_configuration(void) |
| { |
| return FFMPEG_CONFIGURATION; |
| } |
