Add libavresample

This is a new library for audio sample format, channel layout, and sample rate
conversion.

1 files changed, 405 insertions, 0 deletions

diff --git a/libavresample/utils.c b/libavresample/utils.c
new file mode 100644
index 0000000000..f54dcc6ae6
--- /dev/null
+++ b/libavresample/utils.c
@@ -0,0 +1,405 @@
+/*
+ * 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_buffer, 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)
+{
+    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 LIBAV_LICENSE + sizeof(LICENSE_PREFIX) - 1;
+}
+
+const char *avresample_configuration(void)
+{
+    return LIBAV_CONFIGURATION;
+}