forked from len0rd/rockbox
mp3 codec: simpler seeking, and now sets the sample frequency dynamically from the mp3 frame headers rather than from the frequency found by the metadata parser
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7299 a1c6a512-1295-4272-9138-f99709370657
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315304aef6
commit
0da0534d10
1 changed files with 41 additions and 28 deletions
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@ -51,20 +51,38 @@ extern char iramstart[];
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extern char iramend[];
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#endif
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struct codec_api *ci;
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unsigned int samplecount;
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unsigned int samplesdone;
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int stop_skip, start_skip;
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int current_stereo_mode = -1;
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int frequency_divider;
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unsigned int current_frequency = 0;
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void recalc_samplecount(void)
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{
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/* NOTE: currently this doesn't work, the below calculated samples_count
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seems to be right, but sometimes we just don't have all the data we
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need... */
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if (ci->id3->frame_count) {
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/* TODO: 1152 is the frame size in samples for MPEG1 layer 2 and layer 3,
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it's probably not correct at all for MPEG2 and layer 1 */
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samplecount = ci->id3->frame_count*1152 - (start_skip + stop_skip);
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} else {
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samplecount = ci->id3->length * frequency_divider / 10;
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}
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}
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/* this is the codec entry point */
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enum codec_status codec_start(struct codec_api* api)
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{
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struct codec_api *ci = api;
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int Status = 0;
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size_t size;
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int file_end;
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char *InputBuffer;
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unsigned int samplecount;
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unsigned int samplesdone;
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bool first_frame;
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int stop_skip, start_skip;
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int current_stereo_mode = -1;
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int frequency_divider;
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ci = api;
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/* Generic codec inititialisation */
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TEST_CODEC_API(api);
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@ -122,6 +140,7 @@ enum codec_status codec_start(struct codec_api* api)
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frequency_divider = 441;
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ci->configure(DSP_SET_FREQUENCY, (int *)ci->id3->frequency);
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current_frequency = ci->id3->frequency;
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codec_set_replaygain(ci->id3);
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ci->request_buffer(&size, ci->id3->first_frame_offset);
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@ -137,18 +156,7 @@ enum codec_status codec_start(struct codec_api* api)
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start_skip = mpeg_latency[ci->id3->layer];
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}
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/* NOTE: currently this doesn't work, the below calculated samples_count
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seems to be right, but sometimes libmad just can't supply us with
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all the data we need... */
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if (ci->id3->frame_count) {
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/* TODO: 1152 is the frame size in samples for MPEG1 layer 2 and layer 3,
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it's probably not correct at all for MPEG2 and layer 1 */
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samplecount = ci->id3->frame_count*1152 - (start_skip + stop_skip);
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samplesdone = ci->id3->elapsed * frequency_divider / 10;
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} else {
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samplecount = ci->id3->length * frequency_divider / 10;
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samplesdone = ci->id3->elapsed * frequency_divider / 10;
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}
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samplesdone = ci->id3->elapsed * frequency_divider / 10;
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/* This is the decoding loop. */
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while (1) {
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@ -158,21 +166,19 @@ enum codec_status codec_start(struct codec_api* api)
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}
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if (ci->seek_time) {
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unsigned int sample_loc;
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int newpos;
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sample_loc = ci->seek_time * frequency_divider / 10;
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samplesdone = (ci->seek_time-1) * frequency_divider / 10;
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newpos = ci->mp3_get_filepos(ci->seek_time-1);
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if (sample_loc >= samplecount + samplesdone)
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break ;
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if (ci->seek_buffer(newpos)) {
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samplecount += samplesdone - sample_loc;
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samplesdone = sample_loc;
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if (!ci->seek_buffer(newpos)) {
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goto next_track;
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}
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ci->seek_time = 0;
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}
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recalc_samplecount();
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/* Lock buffers */
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if (Stream.error == 0) {
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InputBuffer = ci->request_buffer(&size, INPUT_CHUNK_SIZE);
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@ -224,6 +230,14 @@ enum codec_status codec_start(struct codec_api* api)
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very first frame in the stream. */
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/* TODO: possible for start_skip to exceed one frames worth of samples? */
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if(Frame.header.samplerate != current_frequency) {
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current_frequency = Frame.header.samplerate;
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frequency_divider = current_frequency / 100;
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ci->configure(DSP_SWITCH_FREQUENCY,
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(int *)current_frequency);
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recalc_samplecount();
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}
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if (MAD_NCHANNELS(&Frame.header) == 2) {
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if (current_stereo_mode != STEREO_NONINTERLEAVED) {
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
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@ -248,7 +262,6 @@ enum codec_status codec_start(struct codec_api* api)
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ci->advance_buffer(size);
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samplesdone += Synth.pcm.length;
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samplecount -= Synth.pcm.length;
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ci->set_elapsed(samplesdone / (frequency_divider / 10));
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}
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