forked from len0rd/rockbox
Submit FS#11918: Add 2 more codec types to be able to differentiate between AAC / AAC-HE and MPC SV7 / SV8. Additionally handle ATARI soundfiles in get_codec_base_type() as intended.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29199 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Andree Buschmann
parent
1bb3d61ef3
commit
7345ac124e
7 changed files with 72 additions and 14 deletions
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@ -25,6 +25,8 @@
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#include <stdlib.h>
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#include <ctype.h>
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#include <inttypes.h>
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#define assert(a)
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#include "buffering.h"
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#include "storage.h"
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@ -611,6 +613,16 @@ static inline bool buffer_is_low(void)
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return data_counters.useful < (conf_watermark / 2);
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}
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static uintptr_t beyond_handle(struct memory_handle *h)
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{
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/*
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* the last handle on the chain must leave at least one byte
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* between itself and the first handle, to avoid overflowing the
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* ring by advancing buf_widx up to buf_ridx
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*/
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return h->next != 0 ? ringbuf_offset(h->next) : ringbuf_sub(buf_ridx, 1);
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}
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/* Buffer data for the given handle.
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Return whether or not the buffering should continue explicitly. */
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static bool buffer_handle(int handle_id)
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@ -669,10 +681,10 @@ static bool buffer_handle(int handle_id)
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buffer_len - h->widx);
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ssize_t overlap;
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uintptr_t next_handle = ringbuf_offset(h->next);
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uintptr_t next_handle = beyond_handle(h);
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/* stop copying if it would overwrite the reading position */
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if (ringbuf_add_cross(h->widx, copy_n, buf_ridx) >= 0)
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if (h->widx == next_handle || ringbuf_add_cross(h->widx, copy_n, buf_ridx) >= 0)
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return false;
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/* FIXME: This would overwrite the next handle
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@ -789,8 +801,7 @@ static void rebuffer_handle(int handle_id, size_t newpos)
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LOGFQUEUE("buffering >| Q_RESET_HANDLE %d", handle_id);
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queue_send(&buffering_queue, Q_RESET_HANDLE, handle_id);
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uintptr_t next = ringbuf_offset(h->next);
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if (ringbuf_sub(next, h->data) < h->filesize - newpos)
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if (ringbuf_sub(beyond_handle(h), h->data) < h->filesize - newpos)
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{
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/* There isn't enough space to rebuffer all of the track from its new
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offset, so we ask the user to free some */
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@ -102,16 +102,45 @@ static bool codec_load_next_track(void);
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/** misc external functions */
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/* Used to check whether a new codec must be loaded. See array audio_formats[]
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* in metadata.c */
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int get_codec_base_type(int type)
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{
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int base_type = type;
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switch (type) {
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case AFMT_MPA_L1:
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case AFMT_MPA_L2:
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case AFMT_MPA_L3:
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return AFMT_MPA_L3;
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base_type = AFMT_MPA_L3;
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break;
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case AFMT_MPC_SV7:
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case AFMT_MPC_SV8:
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base_type = AFMT_MPC_SV7;
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break;
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case AFMT_MP4_AAC:
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case AFMT_MP4_AAC_HE:
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base_type = AFMT_MP4_AAC;
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break;
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case AFMT_SAP:
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case AFMT_CMC:
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case AFMT_CM3:
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case AFMT_CMR:
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case AFMT_CMS:
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case AFMT_DMC:
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case AFMT_DLT:
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case AFMT_MPT:
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case AFMT_MPD:
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case AFMT_RMT:
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case AFMT_TMC:
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case AFMT_TM8:
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case AFMT_TM2:
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base_type = AFMT_SAP;
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break;
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default:
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break;
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}
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return type;
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return base_type;
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}
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const char *get_codec_filename(int cod_spec)
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@ -257,7 +257,8 @@ static bool read_chunk_stsd(qtmovie_t *qtmovie, size_t chunk_len)
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stream_skip(qtmovie->stream, entry_remaining);
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} else if (qtmovie->res->format==MAKEFOURCC('m','p','4','a')) {
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if (qtmovie->stream->ci->id3->codectype!=AFMT_MP4_AAC) {
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if (qtmovie->stream->ci->id3->codectype!=AFMT_MP4_AAC &&
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qtmovie->stream->ci->id3->codectype!=AFMT_MP4_AAC_HE) {
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return false;
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}
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@ -86,9 +86,12 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
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/* FLAC */
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[AFMT_FLAC] =
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AFMT_ENTRY("FLAC", "flac", NULL, get_flac_metadata, "flac\0"),
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/* Musepack */
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[AFMT_MPC] =
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AFMT_ENTRY("MPC", "mpc", NULL, get_musepack_metadata,"mpc\0"),
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/* Musepack SV7 */
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[AFMT_MPC_SV7] =
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AFMT_ENTRY("MPCv7", "mpc", NULL, get_musepack_metadata,"mpc\0"),
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/* Musepack SV8 */
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[AFMT_MPC_SV8] =
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AFMT_ENTRY("MPCv8", "mpc", NULL, get_musepack_metadata,"mpc\0"),
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/* A/52 (aka AC3) audio */
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[AFMT_A52] =
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AFMT_ENTRY("AC3", "a52", NULL, get_a52_metadata, "a52\0ac3\0"),
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@ -101,6 +104,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
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/* Advanced Audio Coding in M4A container */
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[AFMT_MP4_AAC] =
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AFMT_ENTRY("AAC", "aac", NULL, get_mp4_metadata, "mp4\0"),
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/* Advanced Audio Coding High Efficiency in M4A container */
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[AFMT_MP4_AAC_HE] =
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AFMT_ENTRY("AAC-HE","aac", NULL, get_mp4_metadata, "mp4\0"),
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/* Shorten */
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[AFMT_SHN] =
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AFMT_ENTRY("SHN","shorten", NULL, get_shn_metadata, "shn\0"),
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@ -46,11 +46,13 @@ enum
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AFMT_PCM_WAV, /* Uncompressed PCM in a WAV file */
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AFMT_OGG_VORBIS, /* Ogg Vorbis */
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AFMT_FLAC, /* FLAC */
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AFMT_MPC, /* Musepack */
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AFMT_MPC_SV7, /* Musepack SV7 */
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AFMT_MPC_SV8, /* Musepack SV8 */
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AFMT_A52, /* A/52 (aka AC3) audio */
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AFMT_WAVPACK, /* WavPack */
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AFMT_MP4_ALAC, /* Apple Lossless Audio Codec */
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AFMT_MP4_AAC, /* Advanced Audio Coding (AAC) in M4A container */
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AFMT_MP4_AAC_HE, /* Advanced Audio Coding (AAC-HE) in M4A container */
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AFMT_SHN, /* Shorten */
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AFMT_SID, /* SID File Format */
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AFMT_ADX, /* ADX File Format */
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@ -300,8 +300,7 @@ static bool read_mp4_esds(int fd, struct mp3entry* id3, uint32_t* size)
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/* Skip 13 bits from above, plus 3 bits, then read 11 bits */
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else if ((length >= 4) && (((bits >> 5) & 0x7ff) == 0x2b7))
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{
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/* extensionAudioObjectType */
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DEBUGF("MP4: extensionAudioType\n");
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/* We found an extensionAudioObjectType */
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type = bits & 0x1f; /* Object type - 5 bits*/
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bits = get_long_be(&buf[4]);
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@ -680,7 +679,6 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
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{
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uint32_t frequency;
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id3->codectype = (type == MP4_mp4a) ? AFMT_MP4_AAC : AFMT_MP4_ALAC;
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lseek(fd, 22, SEEK_CUR);
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read_uint32be(fd, &frequency);
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size -= 26;
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@ -700,11 +698,13 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
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read_mp4_atom(fd, &subsize, &subtype, size);
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size -= 10;
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id3->codectype = AFMT_MP4_AAC;
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if (subtype == MP4_esds)
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{
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sbr_used = read_mp4_esds(fd, id3, &size);
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if (sbr_used)
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{
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id3->codectype = AFMT_MP4_AAC_HE;
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if (SBR_upsampling_used)
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DEBUGF("MP4: AAC-HE, SBR upsampling\n");
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else
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@ -712,6 +712,11 @@ static bool read_mp4_container(int fd, struct mp3entry* id3,
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}
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}
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}
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else
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{
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id3->codectype = AFMT_MP4_ALAC;
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}
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}
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break;
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@ -125,6 +125,8 @@ bool get_musepack_metadata(int fd, struct mp3entry *id3)
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bufused = set_replaygain_sv7(id3, false, header[3], bufused);
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bufused = set_replaygain_sv7(id3, true , header[4], bufused);
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id3->codectype = AFMT_MPC_SV7;
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} else {
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return false; /* only SV7 is allowed within a "MP+" signature */
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}
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@ -191,6 +193,8 @@ bool get_musepack_metadata(int fd, struct mp3entry *id3)
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bufused += set_replaygain_sv8(id3, true , gain, peak, bufused);
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}
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}
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id3->codectype = AFMT_MPC_SV8;
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} else {
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/* No sv8 stream header found */
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return false;
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