diff options
author | Max Kellermann <max@duempel.org> | 2012-03-21 19:25:52 +0100 |
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committer | Max Kellermann <max@duempel.org> | 2012-03-21 19:31:04 +0100 |
commit | 8c5ebdff360021a25b43418d3cd60ea975f5e245 (patch) | |
tree | 5780d19719f2493741fd11777bc39baf5361935b /src | |
parent | 1c84f324a13bcc3d7fbd84f8d59398b1c801247a (diff) | |
download | mpd-8c5ebdff360021a25b43418d3cd60ea975f5e245.tar.gz mpd-8c5ebdff360021a25b43418d3cd60ea975f5e245.tar.xz mpd-8c5ebdff360021a25b43418d3cd60ea975f5e245.zip |
audio_format: remove the reverse_endian attribute
Eliminate support for reverse endian samples from the MPD core. This
moves a lot of complexity to the plugins that really need it (only
ALSA and CDIO currently).
Diffstat (limited to 'src')
-rw-r--r-- | src/audio_format.c | 9 | ||||
-rw-r--r-- | src/audio_format.h | 12 | ||||
-rw-r--r-- | src/decoder_api.c | 3 | ||||
-rw-r--r-- | src/filter/convert_filter_plugin.c | 1 | ||||
-rw-r--r-- | src/filter/normalize_filter_plugin.c | 1 | ||||
-rw-r--r-- | src/filter/replay_gain_filter_plugin.c | 2 | ||||
-rw-r--r-- | src/filter/volume_filter_plugin.c | 2 | ||||
-rw-r--r-- | src/output/alsa_output_plugin.c | 4 | ||||
-rw-r--r-- | src/pcm_convert.c | 41 | ||||
-rw-r--r-- | src/pcm_convert.h | 3 | ||||
-rw-r--r-- | src/pcm_format.c | 2 | ||||
-rw-r--r-- | src/pcm_pack.c | 42 | ||||
-rw-r--r-- | src/pcm_pack.h | 8 |
13 files changed, 20 insertions, 110 deletions
diff --git a/src/audio_format.c b/src/audio_format.c index 06dee8e0f..5a12a299c 100644 --- a/src/audio_format.c +++ b/src/audio_format.c @@ -22,12 +22,6 @@ #include <assert.h> #include <stdio.h> -#if G_BYTE_ORDER == G_BIG_ENDIAN -#define REVERSE_ENDIAN_SUFFIX "_le" -#else -#define REVERSE_ENDIAN_SUFFIX "_be" -#endif - void audio_format_mask_apply(struct audio_format *af, const struct audio_format *mask) @@ -100,9 +94,8 @@ audio_format_to_string(const struct audio_format *af, assert(af != NULL); assert(s != NULL); - snprintf(s->buffer, sizeof(s->buffer), "%u:%s%s:%u", + snprintf(s->buffer, sizeof(s->buffer), "%u:%s:%u", af->sample_rate, sample_format_to_string(af->format), - af->reverse_endian ? REVERSE_ENDIAN_SUFFIX : "", af->channels); return s->buffer; diff --git a/src/audio_format.h b/src/audio_format.h index 293dee1c1..fc4b85880 100644 --- a/src/audio_format.h +++ b/src/audio_format.h @@ -88,13 +88,6 @@ struct audio_format { * fully supported currently. */ uint8_t channels; - - /** - * If zero, then samples are stored in host byte order. If - * nonzero, then samples are stored in the reverse host byte - * order. - */ - bool reverse_endian; }; /** @@ -113,7 +106,6 @@ static inline void audio_format_clear(struct audio_format *af) af->sample_rate = 0; af->format = SAMPLE_FORMAT_UNDEFINED; af->channels = 0; - af->reverse_endian = false; } /** @@ -127,7 +119,6 @@ static inline void audio_format_init(struct audio_format *af, af->sample_rate = sample_rate; af->format = (uint8_t)format; af->channels = channels; - af->reverse_endian = false; } /** @@ -239,8 +230,7 @@ static inline bool audio_format_equals(const struct audio_format *a, { return a->sample_rate == b->sample_rate && a->format == b->format && - a->channels == b->channels && - a->reverse_endian == b->reverse_endian; + a->channels == b->channels; } void diff --git a/src/decoder_api.c b/src/decoder_api.c index 8f12d017a..a45d0f1e6 100644 --- a/src/decoder_api.c +++ b/src/decoder_api.c @@ -56,9 +56,6 @@ decoder_initialized(struct decoder *decoder, dc->in_audio_format = *audio_format; getOutputAudioFormat(audio_format, &dc->out_audio_format); - /* force host byte order, even if the decoder supplies reverse - endian */ - dc->out_audio_format.reverse_endian = false; dc->seekable = seekable; dc->total_time = total_time; diff --git a/src/filter/convert_filter_plugin.c b/src/filter/convert_filter_plugin.c index 7acfc3055..c55b69af2 100644 --- a/src/filter/convert_filter_plugin.c +++ b/src/filter/convert_filter_plugin.c @@ -141,7 +141,6 @@ convert_filter_set(struct filter *_filter, assert(audio_format_valid(&filter->out_audio_format)); assert(out_audio_format != NULL); assert(audio_format_valid(out_audio_format)); - assert(!filter->in_audio_format.reverse_endian); filter->out_audio_format = *out_audio_format; } diff --git a/src/filter/normalize_filter_plugin.c b/src/filter/normalize_filter_plugin.c index fa992f0d4..2151482e4 100644 --- a/src/filter/normalize_filter_plugin.c +++ b/src/filter/normalize_filter_plugin.c @@ -67,7 +67,6 @@ normalize_filter_open(struct filter *_filter, struct normalize_filter *filter = (struct normalize_filter *)_filter; audio_format->format = SAMPLE_FORMAT_S16; - audio_format->reverse_endian = false; filter->compressor = Compressor_new(0); diff --git a/src/filter/replay_gain_filter_plugin.c b/src/filter/replay_gain_filter_plugin.c index c21fe4eaf..583a09f90 100644 --- a/src/filter/replay_gain_filter_plugin.c +++ b/src/filter/replay_gain_filter_plugin.c @@ -140,8 +140,6 @@ replay_gain_filter_open(struct filter *_filter, struct replay_gain_filter *filter = (struct replay_gain_filter *)_filter; - audio_format->reverse_endian = false; - filter->audio_format = *audio_format; pcm_buffer_init(&filter->buffer); diff --git a/src/filter/volume_filter_plugin.c b/src/filter/volume_filter_plugin.c index f87a499ec..e52c0a463 100644 --- a/src/filter/volume_filter_plugin.c +++ b/src/filter/volume_filter_plugin.c @@ -74,8 +74,6 @@ volume_filter_open(struct filter *_filter, struct audio_format *audio_format, { struct volume_filter *filter = (struct volume_filter *)_filter; - audio_format->reverse_endian = false; - filter->audio_format = *audio_format; pcm_buffer_init(&filter->buffer); diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c index c62ad2a46..00312c435 100644 --- a/src/output/alsa_output_plugin.c +++ b/src/output/alsa_output_plugin.c @@ -314,10 +314,8 @@ alsa_output_try_reverse(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, return -EINVAL; int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format); - if (err == 0) { + if (err == 0) audio_format->format = sample_format; - audio_format->reverse_endian = true; - } return err; } diff --git a/src/pcm_convert.c b/src/pcm_convert.c index 4721d2496..fbdfe5e91 100644 --- a/src/pcm_convert.c +++ b/src/pcm_convert.c @@ -21,7 +21,6 @@ #include "pcm_convert.h" #include "pcm_channels.h" #include "pcm_format.h" -#include "pcm_byteswap.h" #include "pcm_pack.h" #include "audio_format.h" #include "glib_compat.h" @@ -45,7 +44,6 @@ void pcm_convert_init(struct pcm_convert_state *state) pcm_buffer_init(&state->format_buffer); pcm_buffer_init(&state->pack_buffer); pcm_buffer_init(&state->channels_buffer); - pcm_buffer_init(&state->byteswap_buffer); } void pcm_convert_deinit(struct pcm_convert_state *state) @@ -56,7 +54,6 @@ void pcm_convert_deinit(struct pcm_convert_state *state) pcm_buffer_deinit(&state->format_buffer); pcm_buffer_deinit(&state->pack_buffer); pcm_buffer_deinit(&state->channels_buffer); - pcm_buffer_deinit(&state->byteswap_buffer); } void @@ -164,11 +161,6 @@ pcm_convert_16(struct pcm_convert_state *state, return NULL; } - if (dest_format->reverse_endian) { - buf = pcm_byteswap_16(&state->byteswap_buffer, buf, len); - assert(buf != NULL); - } - *dest_size_r = len; return buf; } @@ -219,11 +211,6 @@ pcm_convert_24(struct pcm_convert_state *state, return NULL; } - if (dest_format->reverse_endian) { - buf = pcm_byteswap_32(&state->byteswap_buffer, buf, len); - assert(buf != NULL); - } - *dest_size_r = len; return buf; } @@ -261,8 +248,7 @@ pcm_convert_24_packed(struct pcm_convert_state *state, size_t dest_size = num_samples * 3; uint8_t *dest = pcm_buffer_get(&state->pack_buffer, dest_size); - pcm_pack_24(dest, buffer, buffer + num_samples, - dest_format->reverse_endian); + pcm_pack_24(dest, buffer, buffer + num_samples); *dest_size_r = dest_size; return dest; @@ -314,11 +300,6 @@ pcm_convert_32(struct pcm_convert_state *state, return buf; } - if (dest_format->reverse_endian) { - buf = pcm_byteswap_32(&state->byteswap_buffer, buf, len); - assert(buf != NULL); - } - *dest_size_r = len; return buf; } @@ -335,12 +316,6 @@ pcm_convert_float(struct pcm_convert_state *state, assert(dest_format->format == SAMPLE_FORMAT_FLOAT); - if (src_format->reverse_endian || dest_format->reverse_endian) { - g_set_error_literal(error_r, pcm_convert_quark(), 0, - "Reverse endian not supported"); - return NULL; - } - /* convert channels first, hoping the source format is supported (float is not) */ @@ -392,20 +367,6 @@ pcm_convert(struct pcm_convert_state *state, size_t *dest_size_r, GError **error_r) { - if (src_format->reverse_endian) { - /* convert to host byte order, because all of our - conversion libraries assume host byte order */ - - src = pcm_byteswap(&state->byteswap_buffer, src_format->format, - src, src_size); - if (src == NULL) { - g_set_error(error_r, pcm_convert_quark(), 0, - "PCM byte order change of format '%s' is not implemented", - sample_format_to_string(src_format->format)); - return NULL; - } - } - struct audio_format float_format; if (src_format->format == SAMPLE_FORMAT_DSD) { size_t f_size; diff --git a/src/pcm_convert.h b/src/pcm_convert.h index 0a0f85681..2e69013ef 100644 --- a/src/pcm_convert.h +++ b/src/pcm_convert.h @@ -47,9 +47,6 @@ struct pcm_convert_state { /** the buffer for converting the channel count */ struct pcm_buffer channels_buffer; - - /** the buffer for swapping the byte order */ - struct pcm_buffer byteswap_buffer; }; static inline GQuark diff --git a/src/pcm_format.c b/src/pcm_format.c index 31de2200e..f6d6011d6 100644 --- a/src/pcm_format.c +++ b/src/pcm_format.c @@ -76,7 +76,7 @@ pcm_allocate_24_to_24p32(struct pcm_buffer *buffer, const uint8_t *src, int32_t *dest; *dest_size_r = src_size / 3 * sizeof(*dest); dest = pcm_buffer_get(buffer, *dest_size_r); - pcm_unpack_24(dest, src, pcm_end_pointer(src, src_size), false); + pcm_unpack_24(dest, src, pcm_end_pointer(src, src_size)); return dest; } diff --git a/src/pcm_pack.c b/src/pcm_pack.c index 94fda5910..aa241f437 100644 --- a/src/pcm_pack.c +++ b/src/pcm_pack.c @@ -22,11 +22,11 @@ #include <glib.h> static void -pack_sample(uint8_t *dest, const int32_t *src0, bool reverse_endian) +pack_sample(uint8_t *dest, const int32_t *src0) { const uint8_t *src = (const uint8_t *)src0; - if ((G_BYTE_ORDER == G_BIG_ENDIAN) != reverse_endian) + if (G_BYTE_ORDER == G_BIG_ENDIAN) ++src; *dest++ = *src++; @@ -35,31 +35,23 @@ pack_sample(uint8_t *dest, const int32_t *src0, bool reverse_endian) } void -pcm_pack_24(uint8_t *dest, const int32_t *src, const int32_t *src_end, - bool reverse_endian) +pcm_pack_24(uint8_t *dest, const int32_t *src, const int32_t *src_end) { /* duplicate loop to help the compiler's optimizer (constant parameter to the pack_sample() inline function) */ - if (G_LIKELY(!reverse_endian)) { - while (src < src_end) { - pack_sample(dest, src++, false); - dest += 3; - } - } else { - while (src < src_end) { - pack_sample(dest, src++, true); - dest += 3; - } + while (src < src_end) { + pack_sample(dest, src++); + dest += 3; } } static void -unpack_sample(int32_t *dest0, const uint8_t *src, bool reverse_endian) +unpack_sample(int32_t *dest0, const uint8_t *src) { uint8_t *dest = (uint8_t *)dest0; - if ((G_BYTE_ORDER == G_BIG_ENDIAN) != reverse_endian) + if (G_BYTE_ORDER == G_BIG_ENDIAN) /* extend the sign bit to the most fourth byte */ *dest++ = *src & 0x80 ? 0xff : 0x00; @@ -67,27 +59,19 @@ unpack_sample(int32_t *dest0, const uint8_t *src, bool reverse_endian) *dest++ = *src++; *dest++ = *src; - if ((G_BYTE_ORDER == G_LITTLE_ENDIAN) != reverse_endian) + if (G_BYTE_ORDER != G_LITTLE_ENDIAN) /* extend the sign bit to the most fourth byte */ *dest++ = *src & 0x80 ? 0xff : 0x00; } void -pcm_unpack_24(int32_t *dest, const uint8_t *src, const uint8_t *src_end, - bool reverse_endian) +pcm_unpack_24(int32_t *dest, const uint8_t *src, const uint8_t *src_end) { /* duplicate loop to help the compiler's optimizer (constant parameter to the unpack_sample() inline function) */ - if (G_LIKELY(!reverse_endian)) { - while (src < src_end) { - unpack_sample(dest++, src, false); - src += 3; - } - } else { - while (src < src_end) { - unpack_sample(dest++, src, true); - src += 3; - } + while (src < src_end) { + unpack_sample(dest++, src); + src += 3; } } diff --git a/src/pcm_pack.h b/src/pcm_pack.h index 6f275608c..f3184b403 100644 --- a/src/pcm_pack.h +++ b/src/pcm_pack.h @@ -37,11 +37,9 @@ * @param dest the destination buffer (array of triples) * @param src the source buffer * @param num_samples the number of samples to convert - * @param reverse_endian is src and dest in non-host byte order? */ void -pcm_pack_24(uint8_t *dest, const int32_t *src, const int32_t *src_end, - bool reverse_endian); +pcm_pack_24(uint8_t *dest, const int32_t *src, const int32_t *src_end); /** * Converts packed 24 bit samples (3 bytes per sample) to padded 24 @@ -50,10 +48,8 @@ pcm_pack_24(uint8_t *dest, const int32_t *src, const int32_t *src_end, * @param dest the destination buffer * @param src the source buffer (array of triples) * @param num_samples the number of samples to convert - * @param reverse_endian is src and dest in non-host byte order? */ void -pcm_unpack_24(int32_t *dest, const uint8_t *src, const uint8_t *src_end, - bool reverse_endian); +pcm_unpack_24(int32_t *dest, const uint8_t *src, const uint8_t *src_end); #endif |