Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rtmp: Return a proper error code instead of -1
  rtmp: Check malloc calls
  rtmp: Check ff_rtmp_packet_create calls
  lavfi: add audio mix filter
  flvdec: Make sure sample_rate is set to the updated value
  tqi: Pass errors from the MB decoder

Conflicts:
	Changelog
	doc/filters.texi
	libavcodec/eatqi.c
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>

11 files changed, 795 insertions, 86 deletions

diff --git a/Changelog b/Changelog
index 8f6600b485..645449f5d2 100644
--- a/Changelog
+++ b/Changelog
@@ -38,6 +38,7 @@ version 0.11:
 - audio split filter
 - vorbis parser
 - png parser
+- audio mix filter
 
 
 version 0.10:
diff --git a/doc/filters.texi b/doc/filters.texi
index 28d758c923..324a1542e8 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -216,6 +216,44 @@ amovie=input.mkv:si=5 [a5];
 [x3][a5] amerge" -c:a pcm_s16le output.mkv
 @end example
 
+@section amix
+
+Mixes multiple audio inputs into a single output.
+
+For example
+@example
+ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
+@end example
+will mix 3 input audio streams to a single output with the same duration as the
+first input and a dropout transition time of 3 seconds.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item inputs
+Number of inputs. If unspecified, it defaults to 2.
+
+@item duration
+How to determine the end-of-stream.
+@table @option
+
+@item longest
+Duration of longest input. (default)
+
+@item shortest
+Duration of shortest input.
+
+@item first
+Duration of first input.
+
+@end table
+
+@item dropout_transition
+Transition time, in seconds, for volume renormalization when an input
+stream ends. The default value is 2 seconds.
+
+@end table
+
 @section anull
 
 Pass the audio source unchanged to the output.
diff --git a/libavcodec/eatqi.c b/libavcodec/eatqi.c
index 9e773bcb0e..55c080dea8 100644
--- a/libavcodec/eatqi.c
+++ b/libavcodec/eatqi.c
@@ -62,7 +62,7 @@ static int tqi_decode_mb(MpegEncContext *s, DCTELEM (*block)[64])
     int n;
     s->dsp.clear_blocks(block[0]);
     for (n=0; n<6; n++)
-        if(ff_mpeg1_decode_block_intra(s, block[n], n)<0)
+        if (ff_mpeg1_decode_block_intra(s, block[n], n) < 0)
             return -1;
 
     return 0;
@@ -138,7 +138,7 @@ static int tqi_decode_frame(AVCodecContext *avctx,
     for (s->mb_y=0; s->mb_y<(avctx->height+15)/16; s->mb_y++)
     for (s->mb_x=0; s->mb_x<(avctx->width+15)/16; s->mb_x++)
     {
-        if(tqi_decode_mb(s, t->block) < 0)
+        if (tqi_decode_mb(s, t->block) < 0)
             goto end;
         tqi_idct_put(t, t->block);
     }
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 1ec6c625f2..29345fc15e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -47,6 +47,7 @@ OBJS-$(CONFIG_SWSCALE)                       += lswsutils.o
 OBJS-$(CONFIG_ACONVERT_FILTER)               += af_aconvert.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
+OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
new file mode 100644
index 0000000000..3399b7c051
--- /dev/null
+++ b/libavfilter/af_amix.c
@@ -0,0 +1,545 @@
+/*
+ * Audio Mix Filter
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Audio Mix Filter
+ *
+ * Mixes audio from multiple sources into a single output. The channel layout,
+ * sample rate, and sample format will be the same for all inputs and the
+ * output.
+ */
+
+#include "libavutil/audioconvert.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+#define INPUT_OFF      0    /**< input has reached EOF */
+#define INPUT_ON       1    /**< input is active */
+#define INPUT_INACTIVE 2    /**< input is on, but is currently inactive */
+
+#define DURATION_LONGEST  0
+#define DURATION_SHORTEST 1
+#define DURATION_FIRST    2
+
+
+typedef struct FrameInfo {
+    int nb_samples;
+    int64_t pts;
+    struct FrameInfo *next;
+} FrameInfo;
+
+/**
+ * Linked list used to store timestamps and frame sizes of all frames in the
+ * FIFO for the first input.
+ *
+ * This is needed to keep timestamps synchronized for the case where multiple
+ * input frames are pushed to the filter for processing before a frame is
+ * requested by the output link.
+ */
+typedef struct FrameList {
+    int nb_frames;
+    int nb_samples;
+    FrameInfo *list;
+    FrameInfo *end;
+} FrameList;
+
+static void frame_list_clear(FrameList *frame_list)
+{
+    if (frame_list) {
+        while (frame_list->list) {
+            FrameInfo *info = frame_list->list;
+            frame_list->list = info->next;
+            av_free(info);
+        }
+        frame_list->nb_frames  = 0;
+        frame_list->nb_samples = 0;
+        frame_list->end        = NULL;
+    }
+}
+
+static int frame_list_next_frame_size(FrameList *frame_list)
+{
+    if (!frame_list->list)
+        return 0;
+    return frame_list->list->nb_samples;
+}
+
+static int64_t frame_list_next_pts(FrameList *frame_list)
+{
+    if (!frame_list->list)
+        return AV_NOPTS_VALUE;
+    return frame_list->list->pts;
+}
+
+static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
+{
+    if (nb_samples >= frame_list->nb_samples) {
+        frame_list_clear(frame_list);
+    } else {
+        int samples = nb_samples;
+        while (samples > 0) {
+            FrameInfo *info = frame_list->list;
+            av_assert0(info != NULL);
+            if (info->nb_samples <= samples) {
+                samples -= info->nb_samples;
+                frame_list->list = info->next;
+                if (!frame_list->list)
+                    frame_list->end = NULL;
+                frame_list->nb_frames--;
+                frame_list->nb_samples -= info->nb_samples;
+                av_free(info);
+            } else {
+                info->nb_samples       -= samples;
+                info->pts              += samples;
+                frame_list->nb_samples -= samples;
+                samples = 0;
+            }
+        }
+    }
+}
+
+static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
+{
+    FrameInfo *info = av_malloc(sizeof(*info));
+    if (!info)
+        return AVERROR(ENOMEM);
+    info->nb_samples = nb_samples;
+    info->pts        = pts;
+    info->next       = NULL;
+
+    if (!frame_list->list) {
+        frame_list->list = info;
+        frame_list->end  = info;
+    } else {
+        av_assert0(frame_list->end != NULL);
+        frame_list->end->next = info;
+        frame_list->end       = info;
+    }
+    frame_list->nb_frames++;
+    frame_list->nb_samples += nb_samples;
+
+    return 0;
+}
+
+
+typedef struct MixContext {
+    const AVClass *class;       /**< class for AVOptions */
+
+    int nb_inputs;              /**< number of inputs */
+    int active_inputs;          /**< number of input currently active */
+    int duration_mode;          /**< mode for determining duration */
+    float dropout_transition;   /**< transition time when an input drops out */
+
+    int nb_channels;            /**< number of channels */
+    int sample_rate;            /**< sample rate */
+    AVAudioFifo **fifos;        /**< audio fifo for each input */
+    uint8_t *input_state;       /**< current state of each input */
+    float *input_scale;         /**< mixing scale factor for each input */
+    float scale_norm;           /**< normalization factor for all inputs */
+    int64_t next_pts;           /**< calculated pts for next output frame */
+    FrameList *frame_list;      /**< list of frame info for the first input */
+} MixContext;
+
+#define OFFSET(x) offsetof(MixContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption options[] = {
+    { "inputs", "Number of inputs.",
+            OFFSET(nb_inputs), AV_OPT_TYPE_INT, { 2 }, 1, 32, A },
+    { "duration", "How to determine the end-of-stream.",
+            OFFSET(duration_mode), AV_OPT_TYPE_INT, { DURATION_LONGEST }, 0,  2, A, "duration" },
+        { "longest",  "Duration of longest input.",  0, AV_OPT_TYPE_CONST, { DURATION_LONGEST  }, INT_MIN, INT_MAX, A, "duration" },
+        { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { DURATION_SHORTEST }, INT_MIN, INT_MAX, A, "duration" },
+        { "first",    "Duration of first input.",    0, AV_OPT_TYPE_CONST, { DURATION_FIRST    }, INT_MIN, INT_MAX, A, "duration" },
+    { "dropout_transition", "Transition time, in seconds, for volume "
+                            "renormalization when an input stream ends.",
+            OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { 2.0 }, 0, INT_MAX, A },
+    { NULL },
+};
+
+static const AVClass amix_class = {
+    .class_name = "amix filter",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+
+/**
+ * Update the scaling factors to apply to each input during mixing.
+ *
+ * This balances the full volume range between active inputs and handles
+ * volume transitions when EOF is encountered on an input but mixing continues
+ * with the remaining inputs.
+ */
+static void calculate_scales(MixContext *s, int nb_samples)
+{
+    int i;
+
+    if (s->scale_norm > s->active_inputs) {
+        s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);
+        s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);
+    }
+
+    for (i = 0; i < s->nb_inputs; i++) {
+        if (s->input_state[i] == INPUT_ON)
+            s->input_scale[i] = 1.0f / s->scale_norm;
+        else
+            s->input_scale[i] = 0.0f;
+    }
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    MixContext *s      = ctx->priv;
+    int i;
+    char buf[64];
+
+    s->sample_rate     = outlink->sample_rate;
+    outlink->time_base = (AVRational){ 1, outlink->sample_rate };
+    s->next_pts        = AV_NOPTS_VALUE;
+
+    s->frame_list = av_mallocz(sizeof(*s->frame_list));
+    if (!s->frame_list)
+        return AVERROR(ENOMEM);
+
+    s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));
+    if (!s->fifos)
+        return AVERROR(ENOMEM);
+
+    s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
+    for (i = 0; i < s->nb_inputs; i++) {
+        s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
+        if (!s->fifos[i])
+            return AVERROR(ENOMEM);
+    }
+
+    s->input_state = av_malloc(s->nb_inputs);
+    if (!s->input_state)
+        return AVERROR(ENOMEM);
+    memset(s->input_state, INPUT_ON, s->nb_inputs);
+    s->active_inputs = s->nb_inputs;
+
+    s->input_scale = av_mallocz(s->nb_inputs * sizeof(*s->input_scale));
+    if (!s->input_scale)
+        return AVERROR(ENOMEM);
+    s->scale_norm = s->active_inputs;
+    calculate_scales(s, 0);
+
+    av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
+
+    av_log(ctx, AV_LOG_VERBOSE,
+           "inputs:%d fmt:%s srate:%"PRId64" cl:%s\n", s->nb_inputs,
+           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
+
+    return 0;
+}
+
+/* TODO: move optimized version from DSPContext to libavutil */
+static void vector_fmac_scalar(float *dst, const float *src, float mul, int len)
+{
+    int i;
+    for (i = 0; i < len; i++)
+        dst[i] += src[i] * mul;
+}
+
+/**
+ * Read samples from the input FIFOs, mix, and write to the output link.
+ */
+static int output_frame(AVFilterLink *outlink, int nb_samples)
+{
+    AVFilterContext *ctx = outlink->src;
+    MixContext      *s = ctx->priv;
+    AVFilterBufferRef *out_buf, *in_buf;
+    int i;
+
+    calculate_scales(s, nb_samples);
+
+    out_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+    if (!out_buf)
+        return AVERROR(ENOMEM);
+
+    in_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+    if (!in_buf)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < s->nb_inputs; i++) {
+        if (s->input_state[i] == INPUT_ON) {
+            av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
+                               nb_samples);
+            vector_fmac_scalar((float *)out_buf->extended_data[0],
+                               (float *) in_buf->extended_data[0],
+                               s->input_scale[i], nb_samples * s->nb_channels);
+        }
+    }
+    avfilter_unref_buffer(in_buf);
+
+    out_buf->pts = s->next_pts;
+    if (s->next_pts != AV_NOPTS_VALUE)
+        s->next_pts += nb_samples;
+
+    ff_filter_samples(outlink, out_buf);
+
+    return 0;
+}
+
+/**
+ * Returns the smallest number of samples available in the input FIFOs other
+ * than that of the first input.
+ */
+static int get_available_samples(MixContext *s)
+{
+    int i;
+    int available_samples = INT_MAX;
+
+    av_assert0(s->nb_inputs > 1);
+
+    for (i = 1; i < s->nb_inputs; i++) {
+        int nb_samples;
+        if (s->input_state[i] == INPUT_OFF)
+            continue;
+        nb_samples = av_audio_fifo_size(s->fifos[i]);
+        available_samples = FFMIN(available_samples, nb_samples);
+    }
+    if (available_samples == INT_MAX)
+        return 0;
+    return available_samples;
+}
+
+/**
+ * Requests a frame, if needed, from each input link other than the first.
+ */
+static int request_samples(AVFilterContext *ctx, int min_samples)
+{
+    MixContext *s = ctx->priv;
+    int i, ret;
+
+    av_assert0(s->nb_inputs > 1);
+
+    for (i = 1; i < s->nb_inputs; i++) {
+        ret = 0;
+        if (s->input_state[i] == INPUT_OFF)
+            continue;
+        while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples)
+            ret = avfilter_request_frame(ctx->inputs[i]);
+        if (ret == AVERROR_EOF) {
+            if (av_audio_fifo_size(s->fifos[i]) == 0) {
+                s->input_state[i] = INPUT_OFF;
+                continue;
+            }
+        } else if (ret)
+            return ret;
+    }
+    return 0;
+}
+
+/**
+ * Calculates the number of active inputs and determines EOF based on the
+ * duration option.
+ *
+ * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
+ */
+static int calc_active_inputs(MixContext *s)
+{
+    int i;
+    int active_inputs = 0;
+    for (i = 0; i < s->nb_inputs; i++)
+        active_inputs += !!(s->input_state[i] != INPUT_OFF);
+    s->active_inputs = active_inputs;
+
+    if (!active_inputs ||
+        (s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) ||
+        (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
+        return AVERROR_EOF;
+    return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    MixContext      *s = ctx->priv;
+    int ret;
+    int wanted_samples, available_samples;
+
+    if (s->input_state[0] == INPUT_OFF) {
+        ret = request_samples(ctx, 1);
+        if (ret < 0)
+            return ret;
+
+        ret = calc_active_inputs(s);
+        if (ret < 0)
+            return ret;
+
+        available_samples = get_available_samples(s);
+        if (!available_samples)
+            return 0;
+
+        return output_frame(outlink, available_samples);
+    }
+
+    if (s->frame_list->nb_frames == 0) {
+        ret = avfilter_request_frame(ctx->inputs[0]);
+        if (ret == AVERROR_EOF) {
+            s->input_state[0] = INPUT_OFF;
+            if (s->nb_inputs == 1)
+                return AVERROR_EOF;
+            else
+                return AVERROR(EAGAIN);
+        } else if (ret)
+            return ret;
+    }
+    av_assert0(s->frame_list->nb_frames > 0);
+
+    wanted_samples = frame_list_next_frame_size(s->frame_list);
+    ret = request_samples(ctx, wanted_samples);
+    if (ret < 0)
+        return ret;
+
+    ret = calc_active_inputs(s);
+    if (ret < 0)
+        return ret;
+
+    if (s->active_inputs > 1) {
+        available_samples = get_available_samples(s);
+        if (!available_samples)
+            return 0;
+        available_samples = FFMIN(available_samples, wanted_samples);
+    } else {
+        available_samples = wanted_samples;
+    }
+
+    s->next_pts = frame_list_next_pts(s->frame_list);
+    frame_list_remove_samples(s->frame_list, available_samples);
+
+    return output_frame(outlink, available_samples);
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    AVFilterContext  *ctx = inlink->dst;
+    MixContext       *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    int i;
+
+    for (i = 0; i < ctx->input_count; i++)
+        if (ctx->inputs[i] == inlink)
+            break;
+    if (i >= ctx->input_count) {
+        av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
+        return;
+    }
+
+    if (i == 0) {
+        int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
+                                   outlink->time_base);
+        frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+    }
+
+    av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
+                        buf->audio->nb_samples);
+
+    avfilter_unref_buffer(buf);
+}
+
+static int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+    MixContext *s = ctx->priv;
+    int i, ret;
+
+    s->class = &amix_class;
+    av_opt_set_defaults(s);
+
+    if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {
+        av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
+        return ret;
+    }
+    av_opt_free(s);
+
+    for (i = 0; i < s->nb_inputs; i++) {
+        char name[32];
+        AVFilterPad pad = { 0 };
+
+        snprintf(name, sizeof(name), "input%d", i);
+        pad.type           = AVMEDIA_TYPE_AUDIO;
+        pad.name           = av_strdup(name);
+        pad.filter_samples = filter_samples;
+
+        avfilter_insert_inpad(ctx, i, &pad);
+    }
+
+    return 0;
+}
+
+static void uninit(AVFilterContext *ctx)
+{
+    int i;
+    MixContext *s = ctx->priv;
+
+    if (s->fifos) {
+        for (i = 0; i < s->nb_inputs; i++)
+            av_audio_fifo_free(s->fifos[i]);
+        av_freep(&s->fifos);
+    }
+    frame_list_clear(s->frame_list);
+    av_freep(&s->frame_list);
+    av_freep(&s->input_state);
+    av_freep(&s->input_scale);
+
+    for (i = 0; i < ctx->input_count; i++)
+        av_freep(&ctx->input_pads[i].name);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    avfilter_add_format(&formats, AV_SAMPLE_FMT_FLT);
+    avfilter_set_common_formats(ctx, formats);
+    ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());
+    ff_set_common_samplerates(ctx, ff_all_samplerates());
+    return 0;
+}
+
+AVFilter avfilter_af_amix = {
+    .name          = "amix",
+    .description   = NULL_IF_CONFIG_SMALL("Audio mixing."),
+    .priv_size     = sizeof(MixContext),
+
+    .init           = init,
+    .uninit         = uninit,
+    .query_formats  = query_formats,
+
+    .inputs    = (const AVFilterPad[]) {{ .name = NULL}},
+    .outputs   = (const AVFilterPad[]) {{ .name          = "default",
+                                          .type          = AVMEDIA_TYPE_AUDIO,
+                                          .config_props  = config_output,
+                                          .request_frame = request_frame },
+                                        { .name = NULL}},
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index aaaef1900c..b9d44f2fdf 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -37,6 +37,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (ACONVERT,    aconvert,    af);
     REGISTER_FILTER (AFORMAT,     aformat,     af);
     REGISTER_FILTER (AMERGE,      amerge,      af);
+    REGISTER_FILTER (AMIX,        amix,        af);
     REGISTER_FILTER (ANULL,       anull,       af);
     REGISTER_FILTER (ARESAMPLE,   aresample,   af);
     REGISTER_FILTER (ASHOWINFO,   ashowinfo,   af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 7d7a5873c4..c44c45ee5e 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,7 +29,7 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  2
-#define LIBAVFILTER_VERSION_MINOR 76
+#define LIBAVFILTER_VERSION_MINOR 77
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
index d4cc7b9a83..e626999519 100644
--- a/libavformat/flvdec.c
+++ b/libavformat/flvdec.c
@@ -591,8 +591,8 @@ static int flv_read_packet(AVFormatContext *s, AVPacket *pkt)
         }
         if(!st->codec->codec_id){
             flv_set_audio_codec(s, st, st->codec, flags & FLV_AUDIO_CODECID_MASK);
-            flv->last_sample_rate = st->codec->sample_rate;
-            flv->last_channels    = st->codec->channels;
+            flv->last_sample_rate = sample_rate = st->codec->sample_rate;
+            flv->last_channels    = channels    = st->codec->channels;
         } else {
             AVCodecContext ctx;
             ctx.sample_rate = sample_rate;
diff --git a/libavformat/rtmppkt.c b/libavformat/rtmppkt.c
index 61e159b06a..bca4a9dab6 100644
--- a/libavformat/rtmppkt.c
+++ b/libavformat/rtmppkt.c
@@ -79,6 +79,7 @@ int ff_rtmp_packet_read(URLContext *h, RTMPPacket *p,
     uint32_t extra = 0;
     enum RTMPPacketType type;
     int size = 0;
+    int ret;
 
     if (ffurl_read(h, &hdr, 1) != 1)
         return AVERROR(EIO);
@@ -129,8 +130,9 @@ int ff_rtmp_packet_read(URLContext *h, RTMPPacket *p,
     if (hdr != RTMP_PS_TWELVEBYTES)
         timestamp += prev_pkt[channel_id].timestamp;
 
-    if (ff_rtmp_packet_create(p, channel_id, type, timestamp, data_size))
-        return -1;
+    if ((ret = ff_rtmp_packet_create(p, channel_id, type, timestamp,
+                                     data_size)) < 0)
+        return ret;
     p->extra = extra;
     // save history
     prev_pkt[channel_id].channel_id = channel_id;
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index 345b07ca5d..fdba429ab2 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -115,12 +115,16 @@ static const uint8_t rtmp_server_key[] = {
 /**
  * Generate 'connect' call and send it to the server.
  */
-static void gen_connect(URLContext *s, RTMPContext *rt)
+static int gen_connect(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 4096)) < 0)
+        return ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
     p = pkt.data;
 
     ff_amf_write_string(&p, "connect");
@@ -165,19 +169,23 @@ static void gen_connect(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate 'releaseStream' call and send it to the server. It should make
  * the server release some channel for media streams.
  */
-static void gen_release_stream(URLContext *s, RTMPContext *rt)
+static int gen_release_stream(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
-                          29 + strlen(rt->playpath));
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 29 + strlen(rt->playpath))) < 0)
+        return ret;
 
     av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
     p = pkt.data;
@@ -188,19 +196,23 @@ static void gen_release_stream(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate 'FCPublish' call and send it to the server. It should make
  * the server preapare for receiving media streams.
  */
-static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
+static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
-                          25 + strlen(rt->playpath));
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 25 + strlen(rt->playpath))) < 0)
+        return ret;
 
     av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
     p = pkt.data;
@@ -211,19 +223,23 @@ static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate 'FCUnpublish' call and send it to the server. It should make
  * the server destroy stream.
  */
-static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
+static int gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
-                          27 + strlen(rt->playpath));
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 27 + strlen(rt->playpath))) < 0)
+        return ret;
 
     av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
     p = pkt.data;
@@ -234,19 +250,25 @@ static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate 'createStream' call and send it to the server. It should make
  * the server allocate some channel for media streams.
  */
-static void gen_create_stream(URLContext *s, RTMPContext *rt)
+static int gen_create_stream(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
     av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
-    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 25)) < 0)
+        return ret;
 
     p = pkt.data;
     ff_amf_write_string(&p, "createStream");
@@ -256,6 +278,8 @@ static void gen_create_stream(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 
@@ -263,13 +287,17 @@ static void gen_create_stream(URLContext *s, RTMPContext *rt)
  * Generate 'deleteStream' call and send it to the server. It should make
  * the server remove some channel for media streams.
  */
-static void gen_delete_stream(URLContext *s, RTMPContext *rt)
+static int gen_delete_stream(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
     av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
-    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 34)) < 0)
+        return ret;
 
     p = pkt.data;
     ff_amf_write_string(&p, "deleteStream");
@@ -279,20 +307,26 @@ static void gen_delete_stream(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate 'play' call and send it to the server, then ping the server
  * to start actual playing.
  */
-static void gen_play(URLContext *s, RTMPContext *rt)
+static int gen_play(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
     av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
-    ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
-                          29 + strlen(rt->playpath));
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 29 + strlen(rt->playpath))) < 0)
+        return ret;
+
     pkt.extra = rt->main_channel_id;
 
     p = pkt.data;
@@ -306,7 +340,9 @@ static void gen_play(URLContext *s, RTMPContext *rt)
     ff_rtmp_packet_destroy(&pkt);
 
     // set client buffer time disguised in ping packet
-    ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
+                                     1, 10)) < 0)
+        return ret;
 
     p = pkt.data;
     bytestream_put_be16(&p, 3);
@@ -315,19 +351,25 @@ static void gen_play(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate 'publish' call and send it to the server.
  */
-static void gen_publish(URLContext *s, RTMPContext *rt)
+static int gen_publish(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
     av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
-    ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
-                          30 + strlen(rt->playpath));
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 30 + strlen(rt->playpath))) < 0)
+        return ret;
+
     pkt.extra = rt->main_channel_id;
 
     p = pkt.data;
@@ -339,48 +381,65 @@ static void gen_publish(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return ret;
 }
 
 /**
  * Generate ping reply and send it to the server.
  */
-static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
+static int gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
+                                     ppkt->timestamp + 1, 6)) < 0)
+        return ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
     p = pkt.data;
     bytestream_put_be16(&p, 7);
     bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate server bandwidth message and send it to the server.
  */
-static void gen_server_bw(URLContext *s, RTMPContext *rt)
+static int gen_server_bw(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW,
+                                     0, 4)) < 0)
+        return ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW, 0, 4);
     p = pkt.data;
     bytestream_put_be32(&p, 2500000);
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 /**
  * Generate check bandwidth message and send it to the server.
  */
-static void gen_check_bw(URLContext *s, RTMPContext *rt)
+static int gen_check_bw(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 21);
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
+                                     0, 21)) < 0)
+        return ret;
 
     p = pkt.data;
     ff_amf_write_string(&p, "_checkbw");
@@ -389,21 +448,29 @@ static void gen_check_bw(URLContext *s, RTMPContext *rt)
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return ret;
 }
 
 /**
  * Generate report on bytes read so far and send it to the server.
  */
-static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
+static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
 {
     RTMPPacket pkt;
     uint8_t *p;
+    int ret;
+
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ,
+                                     ts, 4)) < 0)
+        return ret;
 
-    ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
     p = pkt.data;
     bytestream_put_be32(&p, rt->bytes_read);
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
+
+    return 0;
 }
 
 //TODO: Move HMAC code somewhere. Eventually.
@@ -421,14 +488,16 @@ static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
  * @param keylen digest key length
  * @param dst    buffer where calculated digest will be stored (32 bytes)
  */
-static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
-                             const uint8_t *key, int keylen, uint8_t *dst)
+static int rtmp_calc_digest(const uint8_t *src, int len, int gap,
+                            const uint8_t *key, int keylen, uint8_t *dst)
 {
     struct AVSHA *sha;
     uint8_t hmac_buf[64+32] = {0};
     int i;
 
     sha = av_mallocz(av_sha_size);
+    if (!sha)
+        return AVERROR(ENOMEM);
 
     if (keylen < 64) {
         memcpy(hmac_buf, key, keylen);
@@ -457,6 +526,8 @@ static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
     av_sha_final(sha, dst);
 
     av_free(sha);
+
+    return 0;
 }
 
 /**
@@ -469,14 +540,18 @@ static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
 {
     int i, digest_pos = 0;
+    int ret;
 
     for (i = 8; i < 12; i++)
         digest_pos += buf[i];
     digest_pos = (digest_pos % 728) + 12;
 
-    rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
-                     rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
-                     buf + digest_pos);
+    ret = rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
+                           rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
+                           buf + digest_pos);
+    if (ret < 0)
+        return ret;
+
     return digest_pos;
 }
 
@@ -491,14 +566,18 @@ static int rtmp_validate_digest(uint8_t *buf, int off)
 {
     int i, digest_pos = 0;
     uint8_t digest[32];
+    int ret;
 
     for (i = 0; i < 4; i++)
         digest_pos += buf[i + off];
     digest_pos = (digest_pos % 728) + off + 4;
 
-    rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
-                     rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
-                     digest);
+    ret = rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
+                           rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
+                           digest);
+    if (ret < 0)
+        return ret;
+
     if (!memcmp(digest, buf + digest_pos, 32))
         return digest_pos;
     return 0;
@@ -526,6 +605,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
     int i;
     int server_pos, client_pos;
     uint8_t digest[32];
+    int ret;
 
     av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
 
@@ -534,17 +614,19 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
     for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
         tosend[i] = av_lfg_get(&rnd) >> 24;
     client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
+    if (client_pos < 0)
+        return client_pos;
 
     ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
     i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
     if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
         av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
-        return -1;
+        return AVERROR(EIO);
     }
     i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
     if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
         av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
-        return -1;
+        return AVERROR(EIO);
     }
 
     av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
@@ -552,33 +634,48 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
 
     if (rt->is_input && serverdata[5] >= 3) {
         server_pos = rtmp_validate_digest(serverdata + 1, 772);
+        if (server_pos < 0)
+            return server_pos;
+
         if (!server_pos) {
             server_pos = rtmp_validate_digest(serverdata + 1, 8);
+            if (server_pos < 0)
+                return server_pos;
+
             if (!server_pos) {
                 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
-                return -1;
+                return AVERROR(EIO);
             }
         }
 
-        rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
-                         rtmp_server_key, sizeof(rtmp_server_key),
-                         digest);
-        rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
-                         digest, 32,
-                         digest);
+        ret = rtmp_calc_digest(tosend + 1 + client_pos, 32, 0, rtmp_server_key,
+                               sizeof(rtmp_server_key), digest);
+        if (ret < 0)
+            return ret;
+
+        ret = rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
+                               digest, 32, digest);
+        if (ret < 0)
+            return ret;
+
         if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
             av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
-            return -1;
+            return AVERROR(EIO);
         }
 
         for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
             tosend[i] = av_lfg_get(&rnd) >> 24;
-        rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
-                         rtmp_player_key, sizeof(rtmp_player_key),
-                         digest);
-        rtmp_calc_digest(tosend,  RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
-                         digest, 32,
-                         tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
+        ret = rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
+                               rtmp_player_key, sizeof(rtmp_player_key),
+                               digest);
+        if (ret < 0)
+            return ret;
+
+        ret = rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
+                               digest, 32,
+                               tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
+        if (ret < 0)
+            return ret;
 
         // write reply back to the server
         ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
@@ -599,6 +696,7 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
 {
     int i, t;
     const uint8_t *data_end = pkt->data + pkt->data_size;
+    int ret;
 
 #ifdef DEBUG
     ff_rtmp_packet_dump(s, pkt);
@@ -623,7 +721,8 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
     case RTMP_PT_PING:
         t = AV_RB16(pkt->data);
         if (t == 6)
-            gen_pong(s, rt, pkt);
+            if ((ret = gen_pong(s, rt, pkt)) < 0)
+                return ret;
         break;
     case RTMP_PT_CLIENT_BW:
         if (pkt->data_size < 4) {
@@ -648,14 +747,18 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
             switch (rt->state) {
             case STATE_HANDSHAKED:
                 if (!rt->is_input) {
-                    gen_release_stream(s, rt);
-                    gen_fcpublish_stream(s, rt);
+                    if ((ret = gen_release_stream(s, rt)) < 0)
+                        return ret;
+                    if ((ret = gen_fcpublish_stream(s, rt)) < 0)
+                        return ret;
                     rt->state = STATE_RELEASING;
                 } else {
-                    gen_server_bw(s, rt);
+                    if ((ret = gen_server_bw(s, rt)) < 0)
+                        return ret;
                     rt->state = STATE_CONNECTING;
                 }
-                gen_create_stream(s, rt);
+                if ((ret = gen_create_stream(s, rt)) < 0)
+                    return ret;
                 break;
             case STATE_FCPUBLISH:
                 rt->state = STATE_CONNECTING;
@@ -679,9 +782,11 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
                     rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
                 }
                 if (rt->is_input) {
-                    gen_play(s, rt);
+                    if ((ret = gen_play(s, rt)) < 0)
+                        return ret;
                 } else {
-                    gen_publish(s, rt);
+                    if ((ret = gen_publish(s, rt)) < 0)
+                        return ret;
                 }
                 rt->state = STATE_READY;
                 break;
@@ -711,7 +816,8 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
             if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
             if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
         } else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
-            gen_check_bw(s, rt);
+            if ((ret = gen_check_bw(s, rt)) < 0)
+                return ret;
         }
         break;
     }
@@ -754,14 +860,15 @@ static int get_packet(URLContext *s, int for_header)
         rt->bytes_read += ret;
         if (rt->bytes_read - rt->last_bytes_read > rt->client_report_size) {
             av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
-            gen_bytes_read(s, rt, rpkt.timestamp + 1);
+            if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
+                return ret;
             rt->last_bytes_read = rt->bytes_read;
         }
 
         ret = rtmp_parse_result(s, rt, &rpkt);
         if (ret < 0) {//serious error in current packet
             ff_rtmp_packet_destroy(&rpkt);
-            return -1;
+            return ret;
         }
         if (rt->state == STATE_STOPPED) {
             ff_rtmp_packet_destroy(&rpkt);
@@ -825,20 +932,21 @@ static int get_packet(URLContext *s, int for_header)
 static int rtmp_close(URLContext *h)
 {
     RTMPContext *rt = h->priv_data;
+    int ret = 0;
 
     if (!rt->is_input) {
         rt->flv_data = NULL;
         if (rt->out_pkt.data_size)
             ff_rtmp_packet_destroy(&rt->out_pkt);
         if (rt->state > STATE_FCPUBLISH)
-            gen_fcunpublish_stream(h, rt);
+            ret = gen_fcunpublish_stream(h, rt);
     }
     if (rt->state > STATE_HANDSHAKED)
-        gen_delete_stream(h, rt);
+        ret = gen_delete_stream(h, rt);
 
     av_freep(&rt->flv_data);
     ffurl_close(rt->stream);
-    return 0;
+    return ret;
 }
 
 /**
@@ -868,14 +976,14 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
         port = RTMP_DEFAULT_PORT;
     ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
 
-    if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
-                   &s->interrupt_callback, NULL) < 0) {
+    if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
+                          &s->interrupt_callback, NULL)) < 0) {
         av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
         goto fail;
     }
 
     rt->state = STATE_START;
-    if (rtmp_handshake(s, rt))
+    if ((ret = rtmp_handshake(s, rt)) < 0)
         goto fail;
 
     rt->chunk_size = 128;
@@ -886,8 +994,8 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
 
     rt->app = av_malloc(APP_MAX_LENGTH);
     if (!rt->app) {
-        rtmp_close(s);
-        return AVERROR(ENOMEM);
+        ret = AVERROR(ENOMEM);
+        goto fail;
     }
 
     //extract "app" part from path
@@ -922,8 +1030,8 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
     if (!rt->playpath) {
         rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
         if (!rt->playpath) {
-            rtmp_close(s);
-            return AVERROR(ENOMEM);
+            ret = AVERROR(ENOMEM);
+            goto fail;
         }
 
         if (!strchr(fname, ':') &&
@@ -938,12 +1046,20 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
 
     if (!rt->tcurl) {
         rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
+        if (!rt->tcurl) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
         ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
                     port, "/%s", rt->app);
     }
 
     if (!rt->flashver) {
         rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
+        if (!rt->flashver) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
         if (rt->is_input) {
             snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
                     RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
@@ -960,7 +1076,8 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
 
     av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
            proto, path, rt->app, rt->playpath);
-    gen_connect(s, rt);
+    if ((ret = gen_connect(s, rt)) < 0)
+        goto fail;
 
     do {
         ret = get_packet(s, 1);
@@ -987,7 +1104,7 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
 
 fail:
     rtmp_close(s);
-    return AVERROR(EIO);
+    return ret;
 }
 
 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
@@ -1024,6 +1141,7 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
     int pktsize, pkttype;
     uint32_t ts;
     const uint8_t *buf_temp = buf;
+    int ret;
 
     do {
         if (rt->skip_bytes) {
@@ -1059,7 +1177,10 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
             }
 
             //this can be a big packet, it's better to send it right here
-            ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
+            if ((ret = ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL,
+                                             pkttype, ts, pktsize)) < 0)
+                return ret;
+
             rt->out_pkt.extra = rt->main_channel_id;
             rt->flv_data = rt->out_pkt.data;
 
diff --git a/tests/fate/audio.mak b/tests/fate/audio.mak
index 5cf457ef34..d741faddb2 100644
--- a/tests/fate/audio.mak
+++ b/tests/fate/audio.mak
@@ -39,8 +39,8 @@ fate-nellymoser-aref-encode: $(AREF)
 fate-nellymoser-aref-encode: CMD = enc_dec_pcm flv wav s16le $(REF) -c:a nellymoser
 fate-nellymoser-aref-encode: CMP = stddev
 fate-nellymoser-aref-encode: REF = ./tests/data/acodec-16000-1.ref.wav
-fate-nellymoser-aref-encode: CMP_SHIFT = -1172
-fate-nellymoser-aref-encode: CMP_TARGET = 9617
+fate-nellymoser-aref-encode: CMP_SHIFT = -244
+fate-nellymoser-aref-encode: CMP_TARGET = 9612
 fate-nellymoser-aref-encode: SIZE_TOLERANCE = 268
 
 FATE_SAMPLES_AUDIO += fate-sierra-vmd-audio