lavd/pulse: rename pulse.c into pulse_audio_dec.c
authorLukasz Marek <lukasz.m.luki@gmail.com>
Fri, 4 Oct 2013 12:29:06 +0000 (14:29 +0200)
committerMichael Niedermayer <michaelni@gmx.at>
Tue, 8 Oct 2013 20:28:39 +0000 (22:28 +0200)
This make name of the file more specific
as there is also encoder implementation.

Signed-off-by: Lukasz Marek <lukasz.m.luki@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
libavdevice/Makefile
libavdevice/pulse.c [deleted file]
libavdevice/pulse_audio_dec.c [new file with mode: 0644]

index 2fdc47b..ca2037a 100644 (file)
@@ -30,7 +30,7 @@ OBJS-$(CONFIG_LAVFI_INDEV)               += lavfi.o
 OBJS-$(CONFIG_OPENAL_INDEV)              += openal-dec.o
 OBJS-$(CONFIG_OSS_INDEV)                 += oss_audio.o
 OBJS-$(CONFIG_OSS_OUTDEV)                += oss_audio.o
-OBJS-$(CONFIG_PULSE_INDEV)               += pulse.o
+OBJS-$(CONFIG_PULSE_INDEV)               += pulse_audio_dec.o
 OBJS-$(CONFIG_PULSE_OUTDEV)              += pulse_audio_enc.o
 OBJS-$(CONFIG_SDL_OUTDEV)                += sdl.o
 OBJS-$(CONFIG_SNDIO_INDEV)               += sndio_common.o sndio_dec.o
diff --git a/libavdevice/pulse.c b/libavdevice/pulse.c
deleted file mode 100644 (file)
index b27329a..0000000
+++ /dev/null
@@ -1,190 +0,0 @@
-/*
- * Pulseaudio input
- * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * PulseAudio input using the simple API.
- * @author Luca Barbato <lu_zero@gentoo.org>
- */
-
-#include <pulse/simple.h>
-#include <pulse/rtclock.h>
-#include <pulse/error.h>
-
-#include "libavformat/avformat.h"
-#include "libavformat/internal.h"
-#include "libavutil/opt.h"
-
-#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
-
-typedef struct PulseData {
-    AVClass *class;
-    char *server;
-    char *name;
-    char *stream_name;
-    int  sample_rate;
-    int  channels;
-    int  frame_size;
-    int  fragment_size;
-    pa_simple *s;
-    int64_t pts;
-    int64_t frame_duration;
-} PulseData;
-
-static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
-    switch (codec_id) {
-    case AV_CODEC_ID_PCM_U8:    return PA_SAMPLE_U8;
-    case AV_CODEC_ID_PCM_ALAW:  return PA_SAMPLE_ALAW;
-    case AV_CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
-    case AV_CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
-    case AV_CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
-    case AV_CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
-    case AV_CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
-    case AV_CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
-    case AV_CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
-    case AV_CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
-    case AV_CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
-    default:                 return PA_SAMPLE_INVALID;
-    }
-}
-
-static av_cold int pulse_read_header(AVFormatContext *s)
-{
-    PulseData *pd = s->priv_data;
-    AVStream *st;
-    char *device = NULL;
-    int ret;
-    enum AVCodecID codec_id =
-        s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
-    const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
-                                pd->sample_rate,
-                                pd->channels };
-
-    pa_buffer_attr attr = { -1 };
-
-    st = avformat_new_stream(s, NULL);
-
-    if (!st) {
-        av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
-        return AVERROR(ENOMEM);
-    }
-
-    attr.fragsize = pd->fragment_size;
-
-    if (strcmp(s->filename, "default"))
-        device = s->filename;
-
-    pd->s = pa_simple_new(pd->server, pd->name,
-                          PA_STREAM_RECORD,
-                          device, pd->stream_name, &ss,
-                          NULL, &attr, &ret);
-
-    if (!pd->s) {
-        av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
-               pa_strerror(ret));
-        return AVERROR(EIO);
-    }
-    /* take real parameters */
-    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id    = codec_id;
-    st->codec->sample_rate = pd->sample_rate;
-    st->codec->channels    = pd->channels;
-    avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
-
-    pd->pts = AV_NOPTS_VALUE;
-    pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
-        (pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
-
-    return 0;
-}
-
-static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
-{
-    PulseData *pd  = s->priv_data;
-    int res;
-    pa_usec_t latency;
-
-    if (av_new_packet(pkt, pd->frame_size) < 0) {
-        return AVERROR(ENOMEM);
-    }
-
-    if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
-        av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
-               pa_strerror(res));
-        av_free_packet(pkt);
-        return AVERROR(EIO);
-    }
-
-    if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
-        av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
-               pa_strerror(res));
-        return AVERROR(EIO);
-    }
-
-    if (pd->pts == AV_NOPTS_VALUE) {
-        pd->pts = -latency;
-    }
-
-    pkt->pts = pd->pts;
-
-    pd->pts += pd->frame_duration;
-
-    return 0;
-}
-
-static av_cold int pulse_close(AVFormatContext *s)
-{
-    PulseData *pd = s->priv_data;
-    pa_simple_free(pd->s);
-    return 0;
-}
-
-#define OFFSET(a) offsetof(PulseData, a)
-#define D AV_OPT_FLAG_DECODING_PARAM
-
-static const AVOption options[] = {
-    { "server",        "pulse server name",                              OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },
-    { "name",          "application name",                               OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },
-    { "stream_name",   "stream description",                             OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
-    { "sample_rate",   "sample rate in Hz",                              OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.i64 = 48000},    1, INT_MAX, D },
-    { "channels",      "number of audio channels",                       OFFSET(channels),      AV_OPT_TYPE_INT,    {.i64 = 2},        1, INT_MAX, D },
-    { "frame_size",    "number of bytes per frame",                      OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.i64 = 1024},     1, INT_MAX, D },
-    { "fragment_size", "buffering size, affects latency and cpu usage",  OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.i64 = -1},      -1, INT_MAX, D },
-    { NULL },
-};
-
-static const AVClass pulse_demuxer_class = {
-    .class_name     = "Pulse demuxer",
-    .item_name      = av_default_item_name,
-    .option         = options,
-    .version        = LIBAVUTIL_VERSION_INT,
-};
-
-AVInputFormat ff_pulse_demuxer = {
-    .name           = "pulse",
-    .long_name      = NULL_IF_CONFIG_SMALL("Pulse audio input"),
-    .priv_data_size = sizeof(PulseData),
-    .read_header    = pulse_read_header,
-    .read_packet    = pulse_read_packet,
-    .read_close     = pulse_close,
-    .flags          = AVFMT_NOFILE,
-    .priv_class     = &pulse_demuxer_class,
-};
diff --git a/libavdevice/pulse_audio_dec.c b/libavdevice/pulse_audio_dec.c
new file mode 100644 (file)
index 0000000..b27329a
--- /dev/null
@@ -0,0 +1,190 @@
+/*
+ * Pulseaudio input
+ * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * PulseAudio input using the simple API.
+ * @author Luca Barbato <lu_zero@gentoo.org>
+ */
+
+#include <pulse/simple.h>
+#include <pulse/rtclock.h>
+#include <pulse/error.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/internal.h"
+#include "libavutil/opt.h"
+
+#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
+
+typedef struct PulseData {
+    AVClass *class;
+    char *server;
+    char *name;
+    char *stream_name;
+    int  sample_rate;
+    int  channels;
+    int  frame_size;
+    int  fragment_size;
+    pa_simple *s;
+    int64_t pts;
+    int64_t frame_duration;
+} PulseData;
+
+static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
+    switch (codec_id) {
+    case AV_CODEC_ID_PCM_U8:    return PA_SAMPLE_U8;
+    case AV_CODEC_ID_PCM_ALAW:  return PA_SAMPLE_ALAW;
+    case AV_CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
+    case AV_CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
+    case AV_CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
+    case AV_CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
+    case AV_CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
+    case AV_CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
+    case AV_CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
+    case AV_CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
+    case AV_CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
+    default:                 return PA_SAMPLE_INVALID;
+    }
+}
+
+static av_cold int pulse_read_header(AVFormatContext *s)
+{
+    PulseData *pd = s->priv_data;
+    AVStream *st;
+    char *device = NULL;
+    int ret;
+    enum AVCodecID codec_id =
+        s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
+    const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
+                                pd->sample_rate,
+                                pd->channels };
+
+    pa_buffer_attr attr = { -1 };
+
+    st = avformat_new_stream(s, NULL);
+
+    if (!st) {
+        av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
+        return AVERROR(ENOMEM);
+    }
+
+    attr.fragsize = pd->fragment_size;
+
+    if (strcmp(s->filename, "default"))
+        device = s->filename;
+
+    pd->s = pa_simple_new(pd->server, pd->name,
+                          PA_STREAM_RECORD,
+                          device, pd->stream_name, &ss,
+                          NULL, &attr, &ret);
+
+    if (!pd->s) {
+        av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
+               pa_strerror(ret));
+        return AVERROR(EIO);
+    }
+    /* take real parameters */
+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
+    st->codec->codec_id    = codec_id;
+    st->codec->sample_rate = pd->sample_rate;
+    st->codec->channels    = pd->channels;
+    avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
+
+    pd->pts = AV_NOPTS_VALUE;
+    pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
+        (pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
+
+    return 0;
+}
+
+static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+    PulseData *pd  = s->priv_data;
+    int res;
+    pa_usec_t latency;
+
+    if (av_new_packet(pkt, pd->frame_size) < 0) {
+        return AVERROR(ENOMEM);
+    }
+
+    if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
+        av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
+               pa_strerror(res));
+        av_free_packet(pkt);
+        return AVERROR(EIO);
+    }
+
+    if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
+        av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
+               pa_strerror(res));
+        return AVERROR(EIO);
+    }
+
+    if (pd->pts == AV_NOPTS_VALUE) {
+        pd->pts = -latency;
+    }
+
+    pkt->pts = pd->pts;
+
+    pd->pts += pd->frame_duration;
+
+    return 0;
+}
+
+static av_cold int pulse_close(AVFormatContext *s)
+{
+    PulseData *pd = s->priv_data;
+    pa_simple_free(pd->s);
+    return 0;
+}
+
+#define OFFSET(a) offsetof(PulseData, a)
+#define D AV_OPT_FLAG_DECODING_PARAM
+
+static const AVOption options[] = {
+    { "server",        "pulse server name",                              OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },
+    { "name",          "application name",                               OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },
+    { "stream_name",   "stream description",                             OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
+    { "sample_rate",   "sample rate in Hz",                              OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.i64 = 48000},    1, INT_MAX, D },
+    { "channels",      "number of audio channels",                       OFFSET(channels),      AV_OPT_TYPE_INT,    {.i64 = 2},        1, INT_MAX, D },
+    { "frame_size",    "number of bytes per frame",                      OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.i64 = 1024},     1, INT_MAX, D },
+    { "fragment_size", "buffering size, affects latency and cpu usage",  OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.i64 = -1},      -1, INT_MAX, D },
+    { NULL },
+};
+
+static const AVClass pulse_demuxer_class = {
+    .class_name     = "Pulse demuxer",
+    .item_name      = av_default_item_name,
+    .option         = options,
+    .version        = LIBAVUTIL_VERSION_INT,
+};
+
+AVInputFormat ff_pulse_demuxer = {
+    .name           = "pulse",
+    .long_name      = NULL_IF_CONFIG_SMALL("Pulse audio input"),
+    .priv_data_size = sizeof(PulseData),
+    .read_header    = pulse_read_header,
+    .read_packet    = pulse_read_packet,
+    .read_close     = pulse_close,
+    .flags          = AVFMT_NOFILE,
+    .priv_class     = &pulse_demuxer_class,
+};