Add pass through support for Opus and VP8; Opus format attribute negotiation

This patch adds pass through support for Opus and VP8. That includes:

* Format attribute negotiation for Opus. Note that unlike some other codecs,
  the draft RFC specifies having spaces delimiting the attributes in addition
  to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
  chan_sip, so a small tweak was also included in this patch for that.

* A format attribute negotiation module for Opus, res_format_attr_opus

* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
  than FIR, this really is specific to VP8 at this time.

Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.

Review: https://reviewboard.asterisk.org/r/2723/

(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
  asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

4 files changed, 32 insertions, 1 deletions

diff --git a/main/channel.c b/main/channel.c
index ea538c91f..1abed5e94 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -802,6 +802,8 @@ struct ast_format *ast_best_codec(struct ast_format_cap *cap, struct ast_format
 		AST_FORMAT_SPEEX32,
 		AST_FORMAT_SPEEX16,
 		AST_FORMAT_SPEEX,
+		/*! Opus */
+		AST_FORMAT_OPUS,
 		/*! SILK is pretty awesome. */
 		AST_FORMAT_SILK,
 		/*! CELT supports crazy high sample rates */
diff --git a/main/format.c b/main/format.c
index 2c563c556..07bc38eb7 100644
--- a/main/format.c
+++ b/main/format.c
@@ -430,6 +430,9 @@ uint64_t ast_format_id_to_old_bitfield(enum ast_format_id id)
 	/*! SpeeX Wideband (16kHz) Free Compression */
 	case AST_FORMAT_SPEEX16:
 		return (1ULL << 33);
+	/*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+	case AST_FORMAT_OPUS:
+		return (1ULL << 34);
 	/*! Raw mu-law data (G.711) */
 	case AST_FORMAT_TESTLAW:
 		return (1ULL << 47);
@@ -449,6 +452,9 @@ uint64_t ast_format_id_to_old_bitfield(enum ast_format_id id)
 	/*! MPEG4 Video */
 	case AST_FORMAT_MP4_VIDEO:
 		return (1ULL << 22);
+	/*! VP8 Video */
+	case AST_FORMAT_VP8:
+		return (1ULL << 23);
 
 	/*! JPEG Images */
 	case AST_FORMAT_JPEG:
@@ -532,6 +538,9 @@ struct ast_format *ast_format_from_old_bitfield(struct ast_format *dst, uint64_t
 	/*! SpeeX Wideband (16kHz) Free Compression */
 	case (1ULL << 33):
 		return ast_format_set(dst, AST_FORMAT_SPEEX16, 0);
+	/*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+	case (1ULL << 34):
+		return ast_format_set(dst, AST_FORMAT_OPUS, 0);
 	/*! Raw mu-law data (G.711) */
 	case (1ULL << 47):
 		return ast_format_set(dst, AST_FORMAT_TESTLAW, 0);
@@ -551,6 +560,9 @@ struct ast_format *ast_format_from_old_bitfield(struct ast_format *dst, uint64_t
 	/*! MPEG4 Video */
 	case (1ULL << 22):
 		return ast_format_set(dst, AST_FORMAT_MP4_VIDEO, 0);
+	/*! VP8 Video */
+	case (1ULL << 23):
+		return ast_format_set(dst, AST_FORMAT_VP8, 0);
 
 	/*! JPEG Images */
 	case (1ULL << 16):
@@ -782,6 +794,9 @@ int ast_format_rate(const struct ast_format *format)
 			return samplerate;
 		}
 	}
+	/* Opus */
+	case AST_FORMAT_OPUS:
+		return 48000;
 	default:
 		return 8000;
 	}
@@ -1067,6 +1082,10 @@ static int format_list_init(void)
 	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), "slin48", 48000, "16 bit Signed Linear PCM (48kHz)", 960, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (48kHz) */
 	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR96, 0), "slin96", 96000, "16 bit Signed Linear PCM (96kHz)", 1920, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (96kHz) */
 	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR192, 0), "slin192", 192000, "16 bit Signed Linear PCM (192kHz)", 3840, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (192kHz) */
+	/* Opus (FIXME: real min is 3/5/10, real max is 120...) */
+	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), "opus", 48000, "Opus Codec", 10, 20, 60, 20, 20, 0, 0);   /*!< codec_opus.c */
+	/* VP8 (passthrough) */
+	format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), "vp8", 0, "VP8 Video", 0, 0, 0, 0 ,0 ,0, 0);         /*!< Passthrough support, see format_h263.c */
 
 	return 0;
 }
diff --git a/main/frame.c b/main/frame.c
index 8822261f6..65bfc008f 100644
--- a/main/frame.c
+++ b/main/frame.c
@@ -1097,9 +1097,13 @@ int ast_codec_get_samples(struct ast_frame *f)
 			return 160;
 		}
 	case AST_FORMAT_CELT:
-		/* TODO The assumes 20ms delivery right now, which is incorrect */
+		/* TODO This assumes 20ms delivery right now, which is incorrect */
 		samples = ast_format_rate(&f->subclass.format) / 50;
 		break;
+	case AST_FORMAT_OPUS:
+		/* TODO This assumes 20ms delivery right now, which is incorrect */
+		samples = 960;
+		break;
 	default:
 		ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
 	}
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 8a9dacf33..c7e359795 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1977,6 +1977,9 @@ int ast_rtp_engine_init()
 	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
 	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
 	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
+	/* Opus and VP8 */
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0,  "audio", "opus", 48000);
+	set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0,  "video", "VP8", 90000);
 
 	/* Define the static rtp payload mappings */
 	add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
@@ -2018,6 +2021,9 @@ int ast_rtp_engine_init()
 	add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
 	add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
 	add_static_payload(121, NULL, AST_RTP_CISCO_DTMF);   /* Must be type 121 */
+	/* Opus and VP8 */
+	add_static_payload(100, ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0);
+	add_static_payload(107, ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0);
 
 	return 0;
 }