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author | Max Kellermann <max@duempel.org> | 2009-11-10 21:46:10 +0100 |
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committer | Max Kellermann <max@duempel.org> | 2009-11-10 21:46:10 +0100 |
commit | 2f69831fb8a0ad13236e91b3edbbcd0a7658a5bf (patch) | |
tree | aba3aa076075c56d7909ef0e3d2fb9d2471fdce3 | |
parent | b6a2ffd3d761c05fd7cd7fecfe470307b6a5ed15 (diff) | |
download | mpd-2f69831fb8a0ad13236e91b3edbbcd0a7658a5bf.tar.gz mpd-2f69831fb8a0ad13236e91b3edbbcd0a7658a5bf.tar.xz mpd-2f69831fb8a0ad13236e91b3edbbcd0a7658a5bf.zip |
decoder/flac: use pcm_buffer instead of fixed buffer
This is a great simplification for flac_common_write(), because we can
convert and submit all of the buffer in one turn. No more partial
buffers with complicated formulas.
-rw-r--r-- | src/decoder/_flac_common.c | 64 | ||||
-rw-r--r-- | src/decoder/_flac_common.h | 6 |
2 files changed, 31 insertions, 39 deletions
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c index 578e73263..fca76d11e 100644 --- a/src/decoder/_flac_common.c +++ b/src/decoder/_flac_common.c @@ -31,6 +31,8 @@ void flac_data_init(struct flac_data *data, struct decoder * decoder, struct input_stream *input_stream) { + pcm_buffer_init(&data->buffer); + data->time = 0; data->position = 0; data->bit_rate = 0; @@ -43,6 +45,8 @@ flac_data_init(struct flac_data *data, struct decoder * decoder, void flac_data_deinit(struct flac_data *data) { + pcm_buffer_deinit(&data->buffer); + if (data->replay_gain_info != NULL) replay_gain_info_free(data->replay_gain_info); @@ -333,45 +337,31 @@ FLAC__StreamDecoderWriteStatus flac_common_write(struct flac_data *data, const FLAC__Frame * frame, const FLAC__int32 *const buf[]) { - unsigned int c_samp; - const unsigned sample_format = data->audio_format.bits; - const unsigned int num_channels = frame->header.channels; - const unsigned int bytes_per_sample = - audio_format_sample_size(&data->audio_format); - const unsigned int bytes_per_channel = - bytes_per_sample * frame->header.channels; - const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel; - unsigned int num_samples; enum decoder_command cmd; + size_t buffer_size = frame->header.blocksize * + audio_format_frame_size(&data->audio_format); + void *buffer; + + buffer = pcm_buffer_get(&data->buffer, buffer_size); + + flac_convert(buffer, data->audio_format.channels, + data->audio_format.bits, buf, + 0, frame->header.blocksize); + + cmd = decoder_data(data->decoder, data->input_stream, + buffer, buffer_size, + data->time, data->bit_rate, + data->replay_gain_info); + switch (cmd) { + case DECODE_COMMAND_NONE: + case DECODE_COMMAND_START: + break; + + case DECODE_COMMAND_STOP: + return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - for (c_samp = 0; c_samp < frame->header.blocksize; - c_samp += num_samples) { - num_samples = frame->header.blocksize - c_samp; - if (num_samples > max_samples) - num_samples = max_samples; - - flac_convert(data->chunk, - num_channels, sample_format, buf, - c_samp, c_samp + num_samples); - - cmd = decoder_data(data->decoder, data->input_stream, - data->chunk, - num_samples * bytes_per_channel, - data->time, data->bit_rate, - data->replay_gain_info); - switch (cmd) { - case DECODE_COMMAND_NONE: - case DECODE_COMMAND_START: - break; - - case DECODE_COMMAND_STOP: - return - FLAC__STREAM_DECODER_WRITE_STATUS_ABORT; - - case DECODE_COMMAND_SEEK: - return - FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; - } + case DECODE_COMMAND_SEEK: + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; } return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; diff --git a/src/decoder/_flac_common.h b/src/decoder/_flac_common.h index 0169e3027..0172ed2b1 100644 --- a/src/decoder/_flac_common.h +++ b/src/decoder/_flac_common.h @@ -24,7 +24,8 @@ #ifndef MPD_FLAC_COMMON_H #define MPD_FLAC_COMMON_H -#include "../decoder_api.h" +#include "decoder_api.h" +#include "pcm_buffer.h" #include "config.h" #include <glib.h> @@ -145,7 +146,8 @@ typedef size_t flac_read_status_size_t; #define FLAC_CHUNK_SIZE 4080 struct flac_data { - unsigned char chunk[FLAC_CHUNK_SIZE]; + struct pcm_buffer buffer; + float time; unsigned int bit_rate; struct audio_format audio_format; |