diff --git a/Changelog b/Changelog
index 8f6600b48515efc4aa3134ee2c175c6c665d90f8..645449f5d229fe5159496d496c5385498b7aa8c2 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 28d758c9239204dd646d45dee1b3e811ac8fce15..324a1542e87a1dda93812fdd6b06e6a183fa466e 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 9e773bcb0e4b226c752ba4ea191a4c53f5eb6490..55c080dea88b60963806b21e2e9d556cbaaac8f2 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 1ec6c625f2c2cd8194f0a62e2c3532786a31afb3..29345fc15ec914fb4d350188b089ac39f0ffeaaa 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 0000000000000000000000000000000000000000..3399b7c051e1251eeb7c7feba23a1b7e44b5e9d2
--- /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 aaaef1900c2ffa804999d0e5c6c30b625375efa2..b9d44f2fdf92c8a3e31249a5b650cfbe21f561e8 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 7d7a5873c4050091d30f9c3de4d815b7f969a5a4..c44c45ee5e226ea2cd62108b8472261210b49c4f 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 d4cc7b9a83874d50251cb34fd03fad8722a9a12f..e626999519c773fcfe6f414c45097b4d3fffaf32 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 61e159b06aafd86293456d13c9a36393259887ea..bca4a9dab622a268079d1b2987155cbd50ca0585 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 345b07ca5d319b524423125d145a5f3ef7f4a65f..fdba429ab22918e64581793ba9b7ef9b8103a77b 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 5cf457ef34bfb9eff25b27a1383595aa03d59020..d741faddb2ad184df8357508e323f985b724c77d 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