forked from len0rd/rockbox
libm4a: Fix warnings introduced in 001a338e51
Change-Id: Ia915e6f8babbd71533f22af566e5c45c2b40fbe5
This commit is contained in:
Solomon Peachy
parent
001a338e51
commit
ac82a653bb
3 changed files with 34 additions and 34 deletions
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@ -28,7 +28,7 @@
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CODEC_HEADER
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/* The maximum buffer size handled by faad. 12 bytes are required by libfaad
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* as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered
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* as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered
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* for each frame. */
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#define FAAD_BYTE_BUFFER_SIZE (2048-12)
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@ -62,7 +62,7 @@ enum codec_status codec_run(void)
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int framelength;
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int lead_trim = 0;
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unsigned int frame_samples;
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unsigned int i;
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uint32_t i;
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unsigned char* buffer;
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NeAACDecFrameInfo frame_info;
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NeAACDecHandle decoder;
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@ -129,7 +129,7 @@ enum codec_status codec_run(void)
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#endif
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i = 0;
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if (param) {
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elapsed_time = param;
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action = CODEC_ACTION_SEEK_TIME;
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@ -138,7 +138,7 @@ enum codec_status codec_run(void)
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* upsampling files the resulting sound_samples_done must be expanded
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* by a factor of 2. This is done via using sbr_fac. */
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if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
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&sound_samples_done, (int*) &i, &seek_idx)) {
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&sound_samples_done, &i, &seek_idx)) {
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sound_samples_done *= sbr_fac;
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} else {
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sound_samples_done = 0;
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@ -151,8 +151,8 @@ enum codec_status codec_run(void)
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}
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ci->set_elapsed(elapsed_time);
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if (i == 0)
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if (i == 0)
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{
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lead_trim = ci->id3->lead_trim;
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}
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@ -168,17 +168,17 @@ enum codec_status codec_run(void)
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/* Deal with any pending seek requests */
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if (action == CODEC_ACTION_SEEK_TIME) {
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/* Seek to the desired time position. Important: When seeking in SBR
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* upsampling files the seek_time must be divided by 2 when calling
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* m4a_seek and the resulting sound_samples_done must be expanded
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* upsampling files the seek_time must be divided by 2 when calling
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* m4a_seek and the resulting sound_samples_done must be expanded
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* by a factor 2. This is done via using sbr_fac. */
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if (m4a_seek(&demux_res, &input_stream,
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(uint64_t) param * ci->id3->frequency / sbr_fac / 1000ULL,
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&sound_samples_done, (int*) &i, &seek_idx)) {
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&sound_samples_done, &i, &seek_idx)) {
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sound_samples_done *= sbr_fac;
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elapsed_time = sound_samples_done * 1000LL / ci->id3->frequency;
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ci->set_elapsed(elapsed_time);
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if (i == 0)
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if (i == 0)
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{
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lead_trim = ci->id3->lead_trim;
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}
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@ -190,9 +190,9 @@ enum codec_status codec_run(void)
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action = CODEC_ACTION_NULL;
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/* There can be gaps between chunks, so skip ahead if needed. It
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* doesn't seem to happen much, but it probably means that a
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* doesn't seem to happen much, but it probably means that a
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* "proper" file can have chunks out of order. Why one would want
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* that an good question (but files with gaps do exist, so who
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* that an good question (but files with gaps do exist, so who
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* knows?), so we don't support that - for now, at least.
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*/
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file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);
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@ -219,7 +219,7 @@ enum codec_status codec_run(void)
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/* Output the audio */
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ci->yield();
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frame_samples = frame_info.samples >> 1;
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if (empty_first_frame)
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@ -238,7 +238,7 @@ enum codec_status codec_run(void)
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/* Gather number of samples for the decoded frame. */
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framelength = frame_samples - lead_trim;
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if (i == demux_res.num_sample_byte_sizes - 1)
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{
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// Size of the last frame
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@ -114,10 +114,10 @@ void stream_create(stream_t *stream,struct codec_api* ci)
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/* Check if there is a dedicated byte position contained for the given frame.
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* Return this byte position in case of success or return -1. This allows to
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* skip empty samples.
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* During standard playback the search result (index i) will always increase.
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* skip empty samples.
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* During standard playback the search result (index i) will always increase.
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* Therefor we save this index and let the caller set this value again as start
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* index when calling m4a_check_sample_offset() for the next frame. This
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* index when calling m4a_check_sample_offset() for the next frame. This
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* reduces the overall loop count significantly. */
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int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame, uint32_t *start)
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{
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@ -139,9 +139,9 @@ int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame, uint32_t *st
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/* Seek to desired sound sample location. Return 1 on success (and modify
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* sound_samples_done and current_sample), 0 if failed. */
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unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
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uint64_t sound_sample_loc, uint64_t* sound_samples_done,
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int* current_sample, int* lookup_table_idx)
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unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
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uint64_t sound_sample_loc, uint64_t* sound_samples_done,
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uint32_t* current_sample, uint32_t* lookup_table_idx)
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{
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uint32_t i, sample_i;
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uint32_t time, time_cnt, time_dur;
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@ -257,7 +257,7 @@ unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
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* 1) the lookup_table array contains the file offset for the first sample
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* of each chunk.
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*
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* 2) the time_to_sample array contains the duration (in sound samples)
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* 2) the time_to_sample array contains the duration (in sound samples)
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* of each sample of data.
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*
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* Locate the chunk containing location (using lookup_table), find the first
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@ -265,8 +265,8 @@ unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
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* calculate the sound_samples_done value.
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*/
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unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
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uint32_t file_loc, uint64_t* sound_samples_done,
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int* current_sample, int* lookup_table_idx)
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uint32_t file_loc, uint64_t* sound_samples_done,
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uint32_t* current_sample, uint32_t* lookup_table_idx)
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{
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uint32_t i;
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uint32_t chunk_sample = 0;
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@ -276,7 +276,7 @@ unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
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uint32_t tmp_cnt;
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uint32_t new_pos;
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/* We know the desired byte offset, search for the chunk right before.
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/* We know the desired byte offset, search for the chunk right before.
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* Return the associated sample to this chunk as chunk_sample. */
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for (i=0; i < demux_res->num_lookup_table; ++i)
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{
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@ -287,7 +287,7 @@ unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
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*lookup_table_idx = i;
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chunk_sample = demux_res->lookup_table[i].sample;
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new_pos = demux_res->lookup_table[i].offset;
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/* Get sound sample offset. */
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i = 0;
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time_to_sample_t *tab2 = demux_res->time_to_sample;
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@ -306,12 +306,12 @@ unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
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}
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/* Go to the new file position. */
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if (stream->ci->seek_buffer(new_pos))
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if (stream->ci->seek_buffer(new_pos))
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{
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*sound_samples_done = new_sound_sample;
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*current_sample = chunk_sample;
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return 1;
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}
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}
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return 0;
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}
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@ -45,13 +45,13 @@ typedef struct {
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typedef uint32_t fourcc_t;
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typedef struct
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typedef struct
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{
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uint32_t first_chunk;
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uint32_t num_samples;
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} sample_to_chunk_t;
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typedef struct
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typedef struct
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{
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uint32_t sample_count;
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uint32_t sample_duration;
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@ -73,10 +73,10 @@ typedef struct
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int32_t sample_to_chunk_offset;
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uint32_t num_sample_to_chunks;
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sample_offset_t *lookup_table;
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uint32_t num_lookup_table;
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time_to_sample_t *time_to_sample;
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uint32_t num_time_to_samples;
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@ -130,10 +130,10 @@ int stream_eof(stream_t *stream);
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void stream_create(stream_t *stream,struct codec_api* ci);
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unsigned int get_sample_offset(demux_res_t *demux_res, uint32_t sample);
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unsigned int m4a_seek (demux_res_t* demux_res, stream_t* stream,
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uint64_t sound_sample_loc, uint64_t* sound_samples_done,
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int* current_sample, int* lookup_table_idx);
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uint64_t sound_sample_loc, uint64_t* sound_samples_done,
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uint32_t* current_sample, uint32_t* lookup_table_idx);
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unsigned int m4a_seek_raw (demux_res_t* demux_res, stream_t* stream,
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uint32_t file_loc, uint64_t* sound_samples_done, int* current_sample, int* lookup_table_idx);
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uint32_t file_loc, uint64_t* sound_samples_done, uint32_t* current_sample, uint32_t* lookup_table_idx);
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int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame, uint32_t *start);
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#endif /* STREAM_H */
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