renamed decoder plugin sources
Make it X_decoder_plugin.c.
This commit is contained in:
Max Kellermann
515
src/decoder/faad_decoder_plugin.c
Normal file
515
src/decoder/faad_decoder_plugin.c
Normal file
@@ -0,0 +1,515 @@
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/*
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* Copyright (C) 2003-2010 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "decoder_api.h"
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#include "decoder_buffer.h"
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#include "audio_check.h"
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#define AAC_MAX_CHANNELS 6
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#include <assert.h>
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#include <unistd.h>
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#include <faad.h>
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#include <glib.h>
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#undef G_LOG_DOMAIN
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#define G_LOG_DOMAIN "faad"
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static const unsigned adts_sample_rates[] =
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{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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};
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/**
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* The GLib quark used for errors reported by this plugin.
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*/
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static inline GQuark
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faad_decoder_quark(void)
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{
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return g_quark_from_static_string("faad");
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}
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/**
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* Check whether the buffer head is an AAC frame, and return the frame
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* length. Returns 0 if it is not a frame.
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*/
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static size_t
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adts_check_frame(const unsigned char *data)
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{
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/* check syncword */
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if (!((data[0] == 0xFF) && ((data[1] & 0xF6) == 0xF0)))
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return 0;
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return (((unsigned int)data[3] & 0x3) << 11) |
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(((unsigned int)data[4]) << 3) |
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(data[5] >> 5);
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}
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/**
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* Find the next AAC frame in the buffer. Returns 0 if no frame is
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* found or if not enough data is available.
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*/
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static size_t
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adts_find_frame(struct decoder_buffer *buffer)
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{
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const unsigned char *data, *p;
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size_t length, frame_length;
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bool ret;
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while (true) {
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data = decoder_buffer_read(buffer, &length);
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if (data == NULL || length < 8) {
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/* not enough data yet */
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ret = decoder_buffer_fill(buffer);
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if (!ret)
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/* failed */
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return 0;
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continue;
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}
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/* find the 0xff marker */
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p = memchr(data, 0xff, length);
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if (p == NULL) {
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/* no marker - discard the buffer */
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decoder_buffer_consume(buffer, length);
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continue;
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}
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if (p > data) {
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/* discard data before 0xff */
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decoder_buffer_consume(buffer, p - data);
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continue;
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}
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/* is it a frame? */
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frame_length = adts_check_frame(data);
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if (frame_length == 0) {
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/* it's just some random 0xff byte; discard it
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and continue searching */
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decoder_buffer_consume(buffer, 1);
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continue;
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}
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if (length < frame_length) {
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/* available buffer size is smaller than the
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frame will be - attempt to read more
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data */
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ret = decoder_buffer_fill(buffer);
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if (!ret) {
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/* not enough data; discard this frame
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to prevent a possible buffer
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overflow */
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data = decoder_buffer_read(buffer, &length);
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if (data != NULL)
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decoder_buffer_consume(buffer, length);
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}
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continue;
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}
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/* found a full frame! */
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return frame_length;
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}
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}
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static float
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adts_song_duration(struct decoder_buffer *buffer)
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{
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unsigned int frames, frame_length;
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unsigned sample_rate = 0;
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float frames_per_second;
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/* Read all frames to ensure correct time and bitrate */
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for (frames = 0;; frames++) {
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frame_length = adts_find_frame(buffer);
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if (frame_length == 0)
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break;
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if (frames == 0) {
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const unsigned char *data;
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size_t buffer_length;
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data = decoder_buffer_read(buffer, &buffer_length);
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assert(data != NULL);
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assert(frame_length <= buffer_length);
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sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2];
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}
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decoder_buffer_consume(buffer, frame_length);
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}
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frames_per_second = (float)sample_rate / 1024.0;
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if (frames_per_second <= 0)
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return -1;
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return (float)frames / frames_per_second;
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}
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static float
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faad_song_duration(struct decoder_buffer *buffer, struct input_stream *is)
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{
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size_t fileread;
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size_t tagsize;
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const unsigned char *data;
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size_t length;
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bool success;
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fileread = is->size >= 0 ? is->size : 0;
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decoder_buffer_fill(buffer);
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data = decoder_buffer_read(buffer, &length);
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if (data == NULL)
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return -1;
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tagsize = 0;
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if (length >= 10 && !memcmp(data, "ID3", 3)) {
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/* skip the ID3 tag */
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tagsize = (data[6] << 21) | (data[7] << 14) |
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(data[8] << 7) | (data[9] << 0);
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tagsize += 10;
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success = decoder_buffer_skip(buffer, tagsize) &&
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decoder_buffer_fill(buffer);
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if (!success)
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return -1;
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data = decoder_buffer_read(buffer, &length);
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if (data == NULL)
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return -1;
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}
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if (is->seekable && length >= 2 &&
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data[0] == 0xFF && ((data[1] & 0xF6) == 0xF0)) {
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/* obtain the duration from the ADTS header */
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float song_length = adts_song_duration(buffer);
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input_stream_seek(is, tagsize, SEEK_SET, NULL);
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data = decoder_buffer_read(buffer, &length);
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if (data != NULL)
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decoder_buffer_consume(buffer, length);
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decoder_buffer_fill(buffer);
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return song_length;
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} else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) {
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/* obtain the duration from the ADIF header */
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unsigned bit_rate;
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size_t skip_size = (data[4] & 0x80) ? 9 : 0;
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if (8 + skip_size > length)
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/* not enough data yet; skip parsing this
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header */
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return -1;
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bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
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(data[5 + skip_size] << 11) |
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(data[6 + skip_size] << 3) |
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(data[7 + skip_size] & 0xE0);
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if (fileread != 0 && bit_rate != 0)
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return fileread * 8.0 / bit_rate;
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else
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return fileread;
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} else
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return -1;
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}
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/**
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* Wrapper for faacDecInit() which works around some API
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* inconsistencies in libfaad.
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*/
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static bool
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faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
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struct audio_format *audio_format, GError **error_r)
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{
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union {
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/* deconst hack for libfaad */
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const void *in;
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void *out;
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} u;
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size_t length;
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int32_t nbytes;
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uint32_t sample_rate;
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uint8_t channels;
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#ifdef HAVE_FAAD_LONG
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/* neaacdec.h declares all arguments as "unsigned long", but
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internally expects uint32_t pointers. To avoid gcc
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warnings, use this workaround. */
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unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
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#else
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uint32_t *sample_rate_p = &sample_rate;
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#endif
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u.in = decoder_buffer_read(buffer, &length);
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if (u.in == NULL) {
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g_set_error(error_r, faad_decoder_quark(), 0,
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"Empty file");
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return false;
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}
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nbytes = faacDecInit(decoder, u.out,
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#ifdef HAVE_FAAD_BUFLEN_FUNCS
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length,
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#endif
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sample_rate_p, &channels);
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if (nbytes < 0) {
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g_set_error(error_r, faad_decoder_quark(), 0,
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"Not an AAC stream");
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return false;
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}
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decoder_buffer_consume(buffer, nbytes);
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return audio_format_init_checked(audio_format, sample_rate,
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SAMPLE_FORMAT_S16, channels, error_r);
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}
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/**
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* Wrapper for faacDecDecode() which works around some API
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* inconsistencies in libfaad.
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*/
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static const void *
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faad_decoder_decode(faacDecHandle decoder, struct decoder_buffer *buffer,
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faacDecFrameInfo *frame_info)
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{
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union {
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/* deconst hack for libfaad */
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const void *in;
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void *out;
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} u;
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size_t length;
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void *result;
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u.in = decoder_buffer_read(buffer, &length);
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if (u.in == NULL)
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return NULL;
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result = faacDecDecode(decoder, frame_info,
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u.out
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#ifdef HAVE_FAAD_BUFLEN_FUNCS
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, length
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#endif
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);
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return result;
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}
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/**
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* Get a song file's total playing time in seconds, as a float.
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* Returns 0 if the duration is unknown, and a negative value if the
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* file is invalid.
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*/
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static float
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faad_get_file_time_float(struct input_stream *is)
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{
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struct decoder_buffer *buffer;
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float length;
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faacDecHandle decoder;
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faacDecConfigurationPtr config;
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buffer = decoder_buffer_new(NULL, is,
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FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
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length = faad_song_duration(buffer, is);
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if (length < 0) {
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bool ret;
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struct audio_format audio_format;
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decoder = faacDecOpen();
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config = faacDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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faacDecSetConfiguration(decoder, config);
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decoder_buffer_fill(buffer);
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ret = faad_decoder_init(decoder, buffer, &audio_format, NULL);
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if (ret)
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length = 0;
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faacDecClose(decoder);
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}
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decoder_buffer_free(buffer);
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return length;
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}
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/**
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* Get a song file's total playing time in seconds, as an int.
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* Returns 0 if the duration is unknown, and a negative value if the
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* file is invalid.
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*/
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static int
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faad_get_file_time(struct input_stream *is)
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{
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int file_time = -1;
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float length;
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if ((length = faad_get_file_time_float(is)) >= 0)
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file_time = length + 0.5;
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return file_time;
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}
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static void
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faad_stream_decode(struct decoder *mpd_decoder, struct input_stream *is)
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{
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GError *error = NULL;
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float total_time = 0;
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faacDecHandle decoder;
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struct audio_format audio_format;
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faacDecConfigurationPtr config;
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bool ret;
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uint16_t bit_rate = 0;
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struct decoder_buffer *buffer;
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enum decoder_command cmd;
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buffer = decoder_buffer_new(mpd_decoder, is,
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FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
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total_time = faad_song_duration(buffer, is);
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/* create the libfaad decoder */
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decoder = faacDecOpen();
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config = faacDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
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config->downMatrix = 1;
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#endif
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#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
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config->dontUpSampleImplicitSBR = 0;
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#endif
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faacDecSetConfiguration(decoder, config);
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while (!decoder_buffer_is_full(buffer) &&
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!input_stream_eof(is) &&
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decoder_get_command(mpd_decoder) == DECODE_COMMAND_NONE) {
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adts_find_frame(buffer);
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decoder_buffer_fill(buffer);
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}
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/* initialize it */
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ret = faad_decoder_init(decoder, buffer, &audio_format, &error);
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if (!ret) {
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g_warning("%s", error->message);
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g_error_free(error);
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faacDecClose(decoder);
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return;
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}
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/* initialize the MPD core */
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decoder_initialized(mpd_decoder, &audio_format, false, total_time);
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/* the decoder loop */
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do {
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size_t frame_size;
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const void *decoded;
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faacDecFrameInfo frame_info;
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/* find the next frame */
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frame_size = adts_find_frame(buffer);
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if (frame_size == 0)
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/* end of file */
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break;
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/* decode it */
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decoded = faad_decoder_decode(decoder, buffer, &frame_info);
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if (frame_info.error > 0) {
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g_warning("error decoding AAC stream: %s\n",
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faacDecGetErrorMessage(frame_info.error));
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break;
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}
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if (frame_info.channels != audio_format.channels) {
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g_warning("channel count changed from %u to %u",
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audio_format.channels, frame_info.channels);
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break;
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}
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#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
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if (frame_info.samplerate != audio_format.sample_rate) {
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g_warning("sample rate changed from %u to %lu",
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audio_format.sample_rate,
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(unsigned long)frame_info.samplerate);
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break;
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}
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#endif
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decoder_buffer_consume(buffer, frame_info.bytesconsumed);
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/* update bit rate and position */
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if (frame_info.samples > 0) {
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bit_rate = frame_info.bytesconsumed * 8.0 *
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frame_info.channels * audio_format.sample_rate /
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frame_info.samples / 1000 + 0.5;
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}
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/* send PCM samples to MPD */
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cmd = decoder_data(mpd_decoder, is, decoded,
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(size_t)frame_info.samples * 2,
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bit_rate);
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} while (cmd != DECODE_COMMAND_STOP);
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/* cleanup */
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faacDecClose(decoder);
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}
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static struct tag *
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faad_stream_tag(struct input_stream *is)
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{
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int file_time = faad_get_file_time(is);
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struct tag *tag;
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if (file_time < 0)
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return NULL;
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tag = tag_new();
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tag->time = file_time;
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return tag;
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}
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static const char *const faad_suffixes[] = { "aac", NULL };
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static const char *const faad_mime_types[] = {
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"audio/aac", "audio/aacp", NULL
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};
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const struct decoder_plugin faad_decoder_plugin = {
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.name = "faad",
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.stream_decode = faad_stream_decode,
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.stream_tag = faad_stream_tag,
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.suffixes = faad_suffixes,
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.mime_types = faad_mime_types,
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};
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