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-rw-r--r--src/decoder/plugins/FaadDecoderPlugin.cxx476
1 files changed, 476 insertions, 0 deletions
diff --git a/src/decoder/plugins/FaadDecoderPlugin.cxx b/src/decoder/plugins/FaadDecoderPlugin.cxx
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+++ b/src/decoder/plugins/FaadDecoderPlugin.cxx
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+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "FaadDecoderPlugin.hxx"
+#include "../DecoderAPI.hxx"
+#include "../DecoderBuffer.hxx"
+#include "input/InputStream.hxx"
+#include "CheckAudioFormat.hxx"
+#include "tag/TagHandler.hxx"
+#include "util/ConstBuffer.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "Log.hxx"
+
+#include <neaacdec.h>
+
+#include <assert.h>
+#include <string.h>
+#include <unistd.h>
+
+static const unsigned adts_sample_rates[] =
+ { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+ 16000, 12000, 11025, 8000, 7350, 0, 0, 0
+};
+
+static constexpr Domain faad_decoder_domain("faad_decoder");
+
+/**
+ * Check whether the buffer head is an AAC frame, and return the frame
+ * length. Returns 0 if it is not a frame.
+ */
+static size_t
+adts_check_frame(const unsigned char *data)
+{
+ /* check syncword */
+ if (!((data[0] == 0xFF) && ((data[1] & 0xF6) == 0xF0)))
+ return 0;
+
+ return (((unsigned int)data[3] & 0x3) << 11) |
+ (((unsigned int)data[4]) << 3) |
+ (data[5] >> 5);
+}
+
+/**
+ * Find the next AAC frame in the buffer. Returns 0 if no frame is
+ * found or if not enough data is available.
+ */
+static size_t
+adts_find_frame(DecoderBuffer *buffer)
+{
+ while (true) {
+ auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_need(buffer, 8));
+ if (data.IsNull())
+ /* failed */
+ return 0;
+
+ /* find the 0xff marker */
+ const uint8_t *p = (const uint8_t *)
+ memchr(data.data, 0xff, data.size);
+ if (p == nullptr) {
+ /* no marker - discard the buffer */
+ decoder_buffer_clear(buffer);
+ continue;
+ }
+
+ if (p > data.data) {
+ /* discard data before 0xff */
+ decoder_buffer_consume(buffer, p - data.data);
+ continue;
+ }
+
+ /* is it a frame? */
+ const size_t frame_length = adts_check_frame(data.data);
+ if (frame_length == 0) {
+ /* it's just some random 0xff byte; discard it
+ and continue searching */
+ decoder_buffer_consume(buffer, 1);
+ continue;
+ }
+
+ if (decoder_buffer_need(buffer, frame_length).IsNull()) {
+ /* not enough data; discard this frame to
+ prevent a possible buffer overflow */
+ decoder_buffer_clear(buffer);
+ continue;
+ }
+
+ /* found a full frame! */
+ return frame_length;
+ }
+}
+
+static float
+adts_song_duration(DecoderBuffer *buffer)
+{
+ const InputStream &is = decoder_buffer_get_stream(buffer);
+ const bool estimate = !is.CheapSeeking();
+ if (estimate && !is.KnownSize())
+ return -1;
+
+ unsigned sample_rate = 0;
+
+ /* Read all frames to ensure correct time and bitrate */
+ unsigned frames = 0;
+ for (;; frames++) {
+ const unsigned frame_length = adts_find_frame(buffer);
+ if (frame_length == 0)
+ break;
+
+ if (frames == 0) {
+ auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
+ assert(!data.IsEmpty());
+ assert(frame_length <= data.size);
+
+ sample_rate = adts_sample_rates[(data.data[2] & 0x3c) >> 2];
+ if (sample_rate == 0)
+ break;
+ }
+
+ decoder_buffer_consume(buffer, frame_length);
+
+ if (estimate && frames == 128) {
+ /* if this is a remote file, don't slurp the
+ whole file just for checking the song
+ duration; instead, stop after some time and
+ extrapolate the song duration from what we
+ have until now */
+
+ const auto offset = is.GetOffset()
+ - decoder_buffer_available(buffer);
+ if (offset <= 0)
+ return -1;
+
+ const auto file_size = is.GetSize();
+ frames = (frames * file_size) / offset;
+ break;
+ }
+ }
+
+ if (sample_rate == 0)
+ return -1;
+
+ float frames_per_second = (float)sample_rate / 1024.0;
+ assert(frames_per_second > 0);
+
+ return (float)frames / frames_per_second;
+}
+
+static float
+faad_song_duration(DecoderBuffer *buffer, InputStream &is)
+{
+ auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_need(buffer, 5));
+ if (data.IsNull())
+ return -1;
+
+ size_t tagsize = 0;
+ if (data.size >= 10 && !memcmp(data.data, "ID3", 3)) {
+ /* skip the ID3 tag */
+
+ tagsize = (data.data[6] << 21) | (data.data[7] << 14) |
+ (data.data[8] << 7) | (data.data[9] << 0);
+
+ tagsize += 10;
+
+ if (!decoder_buffer_skip(buffer, tagsize))
+ return -1;
+
+ data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_need(buffer, 5));
+ if (data.IsNull())
+ return -1;
+ }
+
+ if (data.size >= 8 && adts_check_frame(data.data) > 0) {
+ /* obtain the duration from the ADTS header */
+
+ if (!is.IsSeekable())
+ return -1;
+
+ float song_length = adts_song_duration(buffer);
+
+ is.LockSeek(tagsize, IgnoreError());
+
+ decoder_buffer_clear(buffer);
+
+ return song_length;
+ } else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) {
+ /* obtain the duration from the ADIF header */
+
+ if (!is.KnownSize())
+ return -1;
+
+ size_t skip_size = (data.data[4] & 0x80) ? 9 : 0;
+
+ if (8 + skip_size > data.size)
+ /* not enough data yet; skip parsing this
+ header */
+ return -1;
+
+ unsigned bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) |
+ (data.data[5 + skip_size] << 11) |
+ (data.data[6 + skip_size] << 3) |
+ (data.data[7 + skip_size] & 0xE0);
+
+ const auto size = is.GetSize();
+ if (bit_rate != 0)
+ return size * 8.0 / bit_rate;
+ else
+ return size;
+ } else
+ return -1;
+}
+
+static NeAACDecHandle
+faad_decoder_new()
+{
+ const NeAACDecHandle decoder = NeAACDecOpen();
+
+ NeAACDecConfigurationPtr config =
+ NeAACDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+ config->downMatrix = 1;
+ config->dontUpSampleImplicitSBR = 0;
+ NeAACDecSetConfiguration(decoder, config);
+
+ return decoder;
+}
+
+/**
+ * Wrapper for NeAACDecInit() which works around some API
+ * inconsistencies in libfaad.
+ */
+static bool
+faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
+ AudioFormat &audio_format, Error &error)
+{
+ auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
+ if (data.IsEmpty()) {
+ error.Set(faad_decoder_domain, "Empty file");
+ return false;
+ }
+
+ uint8_t channels;
+ uint32_t sample_rate;
+#ifdef HAVE_FAAD_LONG
+ /* neaacdec.h declares all arguments as "unsigned long", but
+ internally expects uint32_t pointers. To avoid gcc
+ warnings, use this workaround. */
+ unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
+#else
+ uint32_t *sample_rate_p = &sample_rate;
+#endif
+ long nbytes = NeAACDecInit(decoder,
+ /* deconst hack, libfaad requires this */
+ const_cast<unsigned char *>(data.data),
+ data.size,
+ sample_rate_p, &channels);
+ if (nbytes < 0) {
+ error.Set(faad_decoder_domain, "Not an AAC stream");
+ return false;
+ }
+
+ decoder_buffer_consume(buffer, nbytes);
+
+ return audio_format_init_checked(audio_format, sample_rate,
+ SampleFormat::S16, channels, error);
+}
+
+/**
+ * Wrapper for NeAACDecDecode() which works around some API
+ * inconsistencies in libfaad.
+ */
+static const void *
+faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer *buffer,
+ NeAACDecFrameInfo *frame_info)
+{
+ auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
+ if (data.IsEmpty())
+ return nullptr;
+
+ return NeAACDecDecode(decoder, frame_info,
+ /* deconst hack, libfaad requires this */
+ const_cast<uint8_t *>(data.data),
+ data.size);
+}
+
+/**
+ * Get a song file's total playing time in seconds, as a float.
+ * Returns 0 if the duration is unknown, and a negative value if the
+ * file is invalid.
+ */
+static float
+faad_get_file_time_float(InputStream &is)
+{
+ DecoderBuffer *buffer =
+ decoder_buffer_new(nullptr, is,
+ FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
+ float length = faad_song_duration(buffer, is);
+
+ if (length < 0) {
+ NeAACDecHandle decoder = faad_decoder_new();
+
+ decoder_buffer_fill(buffer);
+
+ AudioFormat audio_format;
+ if (faad_decoder_init(decoder, buffer, audio_format,
+ IgnoreError()))
+ length = 0;
+
+ NeAACDecClose(decoder);
+ }
+
+ decoder_buffer_free(buffer);
+
+ return length;
+}
+
+/**
+ * Get a song file's total playing time in seconds, as an int.
+ * Returns 0 if the duration is unknown, and a negative value if the
+ * file is invalid.
+ */
+static int
+faad_get_file_time(InputStream &is)
+{
+ float length = faad_get_file_time_float(is);
+ if (length < 0)
+ return -1;
+
+ return int(length + 0.5);
+}
+
+static void
+faad_stream_decode(Decoder &mpd_decoder, InputStream &is,
+ DecoderBuffer *buffer, const NeAACDecHandle decoder)
+{
+ const float total_time = faad_song_duration(buffer, is);
+
+ if (adts_find_frame(buffer) == 0)
+ return;
+
+ /* initialize it */
+
+ Error error;
+ AudioFormat audio_format;
+ if (!faad_decoder_init(decoder, buffer, audio_format, error)) {
+ LogError(error);
+ return;
+ }
+
+ /* initialize the MPD core */
+
+ decoder_initialized(mpd_decoder, audio_format, false, total_time);
+
+ /* the decoder loop */
+
+ DecoderCommand cmd;
+ unsigned bit_rate = 0;
+ do {
+ /* find the next frame */
+
+ const size_t frame_size = adts_find_frame(buffer);
+ if (frame_size == 0)
+ /* end of file */
+ break;
+
+ /* decode it */
+
+ NeAACDecFrameInfo frame_info;
+ const void *const decoded =
+ faad_decoder_decode(decoder, buffer, &frame_info);
+
+ if (frame_info.error > 0) {
+ FormatWarning(faad_decoder_domain,
+ "error decoding AAC stream: %s",
+ NeAACDecGetErrorMessage(frame_info.error));
+ break;
+ }
+
+ if (frame_info.channels != audio_format.channels) {
+ FormatDefault(faad_decoder_domain,
+ "channel count changed from %u to %u",
+ audio_format.channels, frame_info.channels);
+ break;
+ }
+
+ if (frame_info.samplerate != audio_format.sample_rate) {
+ FormatDefault(faad_decoder_domain,
+ "sample rate changed from %u to %lu",
+ audio_format.sample_rate,
+ (unsigned long)frame_info.samplerate);
+ break;
+ }
+
+ decoder_buffer_consume(buffer, frame_info.bytesconsumed);
+
+ /* update bit rate and position */
+
+ if (frame_info.samples > 0) {
+ bit_rate = frame_info.bytesconsumed * 8.0 *
+ frame_info.channels * audio_format.sample_rate /
+ frame_info.samples / 1000 + 0.5;
+ }
+
+ /* send PCM samples to MPD */
+
+ cmd = decoder_data(mpd_decoder, is, decoded,
+ (size_t)frame_info.samples * 2,
+ bit_rate);
+ } while (cmd != DecoderCommand::STOP);
+}
+
+static void
+faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
+{
+ DecoderBuffer *buffer =
+ decoder_buffer_new(&mpd_decoder, is,
+ FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
+
+ /* create the libfaad decoder */
+
+ const NeAACDecHandle decoder = faad_decoder_new();
+
+ faad_stream_decode(mpd_decoder, is, buffer, decoder);
+
+ /* cleanup */
+
+ NeAACDecClose(decoder);
+ decoder_buffer_free(buffer);
+}
+
+static bool
+faad_scan_stream(InputStream &is,
+ const struct tag_handler *handler, void *handler_ctx)
+{
+ int file_time = faad_get_file_time(is);
+ if (file_time < 0)
+ return false;
+
+ tag_handler_invoke_duration(handler, handler_ctx, file_time);
+ return true;
+}
+
+static const char *const faad_suffixes[] = { "aac", nullptr };
+static const char *const faad_mime_types[] = {
+ "audio/aac", "audio/aacp", nullptr
+};
+
+const DecoderPlugin faad_decoder_plugin = {
+ "faad",
+ nullptr,
+ nullptr,
+ faad_stream_decode,
+ nullptr,
+ nullptr,
+ faad_scan_stream,
+ nullptr,
+ faad_suffixes,
+ faad_mime_types,
+};