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Remove trailing whitespace from all files (#497)
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356 changed files with 41701 additions and 41680 deletions
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@ -37,10 +37,10 @@ This library provides several interfaces to load, demux and decode MPEG video
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and audio data. A high-level API combines the demuxer, video & audio decoders
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in an easy to use wrapper.
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Lower-level APIs for accessing the demuxer, video decoder and audio decoder,
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Lower-level APIs for accessing the demuxer, video decoder and audio decoder,
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as well as providing different data sources are also available.
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Interfaces are written in an object orientet style, meaning you create object
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Interfaces are written in an object orientet style, meaning you create object
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instances via various different constructor functions (plm_*create()),
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do some work on them and later dispose them via plm_*destroy().
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@ -51,11 +51,11 @@ plm_video_* -- the MPEG1 Video ("mpeg1") decoder
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plm_audio_* -- the MPEG1 Audio Layer II ("mp2") decoder
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This library uses malloc(), realloc() and free() to manage memory. Typically
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This library uses malloc(), realloc() and free() to manage memory. Typically
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all allocation happens up-front when creating the interface. However, the
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default buffer size may be too small for certain inputs. In these cases plmpeg
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will realloc() the buffer with a larger size whenever needed. You can configure
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the default buffer size by defining PLM_BUFFER_DEFAULT_SIZE *before*
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the default buffer size by defining PLM_BUFFER_DEFAULT_SIZE *before*
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including this library.
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With the high-level interface you have two options to decode video & audio:
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@ -85,7 +85,7 @@ mat4 rec601 = mat4(
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gl_FragColor = vec4(y, cb, cr, 1.0) * rec601;
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Audio data is decoded into a struct with either one single float array with the
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samples for the left and right channel interleaved, or if the
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samples for the left and right channel interleaved, or if the
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PLM_AUDIO_SEPARATE_CHANNELS is defined *before* including this library, into
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two separate float arrays - one for each channel.
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@ -115,8 +115,8 @@ plm_packet_t *plm_demux_decode(plm_demux_t *self) {
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do {
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code = plm_buffer_next_start_code(self->buffer);
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if (
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code == PLM_DEMUX_PACKET_VIDEO_1 ||
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code == PLM_DEMUX_PACKET_PRIVATE ||
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code == PLM_DEMUX_PACKET_VIDEO_1 ||
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code == PLM_DEMUX_PACKET_PRIVATE ||
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(code >= PLM_DEMUX_PACKET_AUDIO_1 && code <= PLM_DEMUX_PACKET_AUDIO_4)
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) {
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return plm_demux_decode_packet(self, code);
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@ -273,7 +273,7 @@ typedef plm_audio_t plm_audio_t;
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int plm_audio_decode_header(plm_audio_t *self);
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void plm_audio_decode_frame(plm_audio_t *self);
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const plm_quantizer_spec_t *plm_audio_read_allocation(plm_audio_t *self, int sb, int tab3);
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void plm_audio_read_samples(plm_audio_t *self, int ch, int sb, int part);
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void plm_audio_read_samples(plm_audio_t *self, int ch, int sb, int part);
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void plm_audio_matrix_transform(int s[32][3], int ss, float *d, int dp);
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plm_audio_t *plm_audio_create_with_buffer(plm_buffer_t *buffer, int destroy_when_done) {
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@ -345,7 +345,7 @@ plm_samples_t *plm_audio_decode(plm_audio_t *self) {
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self->samples.time = self->time;
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self->samples_decoded += PLM_AUDIO_SAMPLES_PER_FRAME;
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self->time = (double)self->samples_decoded /
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self->time = (double)self->samples_decoded /
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(double)PLM_AUDIO_SAMPLE_RATE[self->samplerate_index];
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return &self->samples;
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@ -402,7 +402,7 @@ int plm_audio_decode_header(plm_audio_t *self) {
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plm_buffer_skip(self->buffer, 16);
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}
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// Compute frame size, check if we have enough data to decode the whole
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// Compute frame size, check if we have enough data to decode the whole
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// frame.
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int bitrate = PLM_AUDIO_BIT_RATE[self->bitrate_index];
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int samplerate = PLM_AUDIO_SAMPLE_RATE[self->samplerate_index];
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@ -561,7 +561,7 @@ void plm_audio_decode_frame(plm_audio_t *self) {
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}
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#else
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for (int j = 0; j < 32; j++) {
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self->samples.interleaved[((out_pos + j) << 1) + ch] =
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self->samples.interleaved[((out_pos + j) << 1) + ch] =
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self->U[j] / 2147418112.0f;
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}
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#endif
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@ -94,7 +94,7 @@ plm_t *plm_create_with_buffer(plm_buffer_t *buffer, int destroy_when_done) {
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self->demux = plm_demux_create(buffer, destroy_when_done);
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// In theory we should check plm_demux_get_num_video_streams() and
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// In theory we should check plm_demux_get_num_video_streams() and
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// plm_demux_get_num_audio_streams() here, but older files typically
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// do not specify these correctly. So we just assume we have a video and
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// audio stream and create the decoders.
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@ -159,7 +159,7 @@ double plm_get_framerate(plm_t *self) {
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int plm_get_num_audio_streams(plm_t *self) {
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// Some files do not specify the number of audio streams in the system header.
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// If the reported number of streams is 0, we check if we have a samplerate,
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// If the reported number of streams is 0, we check if we have a samplerate,
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// indicating at least one audio stream.
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int num_streams = plm_demux_get_num_audio_streams(self->demux);
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return num_streams == 0 && plm_get_samplerate(self) ? 1 : num_streams;
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