| /* GStreamer SBC audio encoder |
| * Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org> |
| * Copyright (C) 2013 Tim-Philipp Müller <tim centricular net> |
| * |
| * This library 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. |
| * |
| * This library 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 this library; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
| * |
| */ |
| |
| /** |
| * SECTION:element-sbenc |
| * |
| * This element encodes raw integer PCM audio into a Bluetooth SBC audio. |
| * |
| * Encoding paramets such as blocks, subbands, bitpool, channel-mode, and |
| * allocation-mode can be set by adding a capsfilter element with appropriate |
| * filtercaps after the sbcenc encoder element. |
| * |
| * <refsect2> |
| * <title>Example pipelines</title> |
| * |[ |
| * gst-launch-1.0 -v audiotestsrc ! sbcenc ! rtpsbcpay ! udpsink |
| * ]| Encode a sine wave into SBC, RTP payload it and send over the network using UDP |
| * </refsect2> |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include <config.h> |
| #endif |
| |
| #include <string.h> |
| |
| #include "gstsbcenc.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (sbc_enc_debug); |
| #define GST_CAT_DEFAULT sbc_enc_debug |
| |
| G_DEFINE_TYPE (GstSbcEnc, gst_sbc_enc, GST_TYPE_AUDIO_ENCODER); |
| |
| static GstStaticPadTemplate sbc_enc_sink_factory = |
| GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw, format=" GST_AUDIO_NE (S16) ", " |
| "rate = (int) { 16000, 32000, 44100, 48000 }, " |
| "channels = (int) [ 1, 2 ]")); |
| |
| static GstStaticPadTemplate sbc_enc_src_factory = |
| GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-sbc, " |
| "rate = (int) { 16000, 32000, 44100, 48000 }, " |
| "channels = (int) [ 1, 2 ], " |
| "channel-mode = (string) { mono, dual, stereo, joint }, " |
| "blocks = (int) { 4, 8, 12, 16 }, " |
| "subbands = (int) { 4, 8 }, " |
| "allocation-method = (string) { snr, loudness }, " |
| "bitpool = (int) [ 2, 64 ]")); |
| |
| |
| static gboolean gst_sbc_enc_start (GstAudioEncoder * enc); |
| static gboolean gst_sbc_enc_stop (GstAudioEncoder * enc); |
| static gboolean gst_sbc_enc_set_format (GstAudioEncoder * enc, |
| GstAudioInfo * info); |
| static GstFlowReturn gst_sbc_enc_handle_frame (GstAudioEncoder * enc, |
| GstBuffer * buffer); |
| |
| static gboolean |
| gst_sbc_enc_set_format (GstAudioEncoder * audio_enc, GstAudioInfo * info) |
| { |
| const gchar *allocation_method, *channel_mode; |
| GstSbcEnc *enc = GST_SBC_ENC (audio_enc); |
| GstStructure *s; |
| GstCaps *caps, *filter_caps; |
| GstCaps *output_caps = NULL; |
| guint sampleframes_per_frame; |
| |
| enc->rate = GST_AUDIO_INFO_RATE (info); |
| enc->channels = GST_AUDIO_INFO_CHANNELS (info); |
| |
| /* negotiate output format based on downstream caps restrictions */ |
| caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc)); |
| if (caps == GST_CAPS_NONE || gst_caps_is_empty (caps)) |
| goto failure; |
| |
| if (caps == NULL) |
| caps = gst_static_pad_template_get_caps (&sbc_enc_src_factory); |
| |
| /* fixate output caps */ |
| filter_caps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT, |
| enc->rate, "channels", G_TYPE_INT, enc->channels, NULL); |
| output_caps = gst_caps_intersect (caps, filter_caps); |
| gst_caps_unref (filter_caps); |
| |
| if (output_caps == NULL || gst_caps_is_empty (output_caps)) { |
| GST_WARNING_OBJECT (enc, "Couldn't negotiate output caps with input rate " |
| "%d and input channels %d and allowed output caps %" GST_PTR_FORMAT, |
| enc->rate, enc->channels, caps); |
| goto failure; |
| } |
| |
| gst_caps_unref (caps); |
| caps = NULL; |
| |
| GST_DEBUG_OBJECT (enc, "fixating caps %" GST_PTR_FORMAT, output_caps); |
| output_caps = gst_caps_truncate (output_caps); |
| s = gst_caps_get_structure (output_caps, 0); |
| if (enc->channels == 1) |
| gst_structure_fixate_field_string (s, "channel-mode", "mono"); |
| else |
| gst_structure_fixate_field_string (s, "channel-mode", "joint"); |
| |
| gst_structure_fixate_field_nearest_int (s, "bitpool", 64); |
| gst_structure_fixate_field_nearest_int (s, "blocks", 16); |
| gst_structure_fixate_field_nearest_int (s, "subbands", 8); |
| gst_structure_fixate_field_string (s, "allocation-method", "loudness"); |
| s = NULL; |
| |
| /* in case there's anything else left to fixate */ |
| output_caps = gst_caps_fixate (output_caps); |
| gst_caps_set_simple (output_caps, "parsed", G_TYPE_BOOLEAN, TRUE, NULL); |
| |
| GST_INFO_OBJECT (enc, "output caps %" GST_PTR_FORMAT, output_caps); |
| |
| /* let's see what we fixated to */ |
| s = gst_caps_get_structure (output_caps, 0); |
| gst_structure_get_int (s, "blocks", &enc->blocks); |
| gst_structure_get_int (s, "subbands", &enc->subbands); |
| gst_structure_get_int (s, "bitpool", &enc->bitpool); |
| allocation_method = gst_structure_get_string (s, "allocation-method"); |
| channel_mode = gst_structure_get_string (s, "channel-mode"); |
| |
| /* We want channel-mode and channels coherent */ |
| if (enc->channels == 1) { |
| if (g_strcmp0 (channel_mode, "mono") != 0) { |
| GST_ERROR_OBJECT (enc, "Can't have channel-mode '%s' for 1 channel", |
| channel_mode); |
| goto failure; |
| } |
| } else { |
| if (g_strcmp0 (channel_mode, "joint") != 0 && |
| g_strcmp0 (channel_mode, "stereo") != 0 && |
| g_strcmp0 (channel_mode, "dual") != 0) { |
| GST_ERROR_OBJECT (enc, "Can't have channel-mode '%s' for 2 channels", |
| channel_mode); |
| goto failure; |
| } |
| } |
| |
| /* we want to be handed all available samples in handle_frame, but always |
| * enough to encode a frame */ |
| sampleframes_per_frame = enc->blocks * enc->subbands; |
| gst_audio_encoder_set_frame_samples_min (audio_enc, sampleframes_per_frame); |
| gst_audio_encoder_set_frame_samples_max (audio_enc, sampleframes_per_frame); |
| gst_audio_encoder_set_frame_max (audio_enc, 0); |
| |
| /* FIXME: what to do with left-over samples at the end? can we encode them? */ |
| gst_audio_encoder_set_hard_min (audio_enc, TRUE); |
| |
| /* and configure encoder based on the output caps we negotiated */ |
| if (enc->rate == 16000) |
| enc->sbc.frequency = SBC_FREQ_16000; |
| else if (enc->rate == 32000) |
| enc->sbc.frequency = SBC_FREQ_32000; |
| else if (enc->rate == 44100) |
| enc->sbc.frequency = SBC_FREQ_44100; |
| else if (enc->rate == 48000) |
| enc->sbc.frequency = SBC_FREQ_48000; |
| else |
| goto failure; |
| |
| if (enc->blocks == 4) |
| enc->sbc.blocks = SBC_BLK_4; |
| else if (enc->blocks == 8) |
| enc->sbc.blocks = SBC_BLK_8; |
| else if (enc->blocks == 12) |
| enc->sbc.blocks = SBC_BLK_12; |
| else if (enc->blocks == 16) |
| enc->sbc.blocks = SBC_BLK_16; |
| else |
| goto failure; |
| |
| enc->sbc.subbands = (enc->subbands == 4) ? SBC_SB_4 : SBC_SB_8; |
| enc->sbc.bitpool = enc->bitpool; |
| |
| if (channel_mode == NULL || allocation_method == NULL) |
| goto failure; |
| |
| if (strcmp (channel_mode, "joint") == 0) |
| enc->sbc.mode = SBC_MODE_JOINT_STEREO; |
| else if (strcmp (channel_mode, "stereo") == 0) |
| enc->sbc.mode = SBC_MODE_STEREO; |
| else if (strcmp (channel_mode, "dual") == 0) |
| enc->sbc.mode = SBC_MODE_DUAL_CHANNEL; |
| else if (strcmp (channel_mode, "mono") == 0) |
| enc->sbc.mode = SBC_MODE_MONO; |
| else if (strcmp (channel_mode, "auto") == 0) |
| enc->sbc.mode = SBC_MODE_JOINT_STEREO; |
| else |
| goto failure; |
| |
| if (strcmp (allocation_method, "loudness") == 0) |
| enc->sbc.allocation = SBC_AM_LOUDNESS; |
| else if (strcmp (allocation_method, "snr") == 0) |
| enc->sbc.allocation = SBC_AM_SNR; |
| else |
| goto failure; |
| |
| if (!gst_audio_encoder_set_output_format (audio_enc, output_caps)) |
| goto failure; |
| |
| return gst_audio_encoder_negotiate (audio_enc); |
| |
| failure: |
| if (output_caps) |
| gst_caps_unref (output_caps); |
| if (caps) |
| gst_caps_unref (caps); |
| return FALSE; |
| } |
| |
| static GstFlowReturn |
| gst_sbc_enc_handle_frame (GstAudioEncoder * audio_enc, GstBuffer * buffer) |
| { |
| GstSbcEnc *enc = GST_SBC_ENC (audio_enc); |
| GstMapInfo in_map, out_map; |
| GstBuffer *outbuf = NULL; |
| guint samples_per_frame, frames, i = 0; |
| |
| /* no fancy draining */ |
| if (buffer == NULL) |
| return GST_FLOW_OK; |
| |
| if (G_UNLIKELY (enc->channels == 0 || enc->blocks == 0 || enc->subbands == 0)) |
| return GST_FLOW_NOT_NEGOTIATED; |
| |
| samples_per_frame = enc->channels * enc->blocks * enc->subbands; |
| |
| if (!gst_buffer_map (buffer, &in_map, GST_MAP_READ)) |
| goto map_failed; |
| |
| frames = in_map.size / (samples_per_frame * sizeof (gint16)); |
| |
| GST_LOG_OBJECT (enc, |
| "encoding %" G_GSIZE_FORMAT " samples into %u SBC frames", |
| in_map.size / (enc->channels * sizeof (gint16)), frames); |
| |
| if (frames > 0) { |
| gsize frame_len; |
| |
| frame_len = sbc_get_frame_length (&enc->sbc); |
| outbuf = gst_audio_encoder_allocate_output_buffer (audio_enc, |
| frames * frame_len); |
| |
| if (outbuf == NULL) |
| goto no_buffer; |
| |
| gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE); |
| |
| for (i = 0; i < frames; ++i) { |
| gssize ret, written = 0; |
| |
| ret = sbc_encode (&enc->sbc, in_map.data + (i * samples_per_frame * 2), |
| samples_per_frame * 2, out_map.data + (i * frame_len), frame_len, |
| &written); |
| |
| if (ret < 0 || written != frame_len) { |
| GST_WARNING_OBJECT (enc, "encoding error, ret = %" G_GSSIZE_FORMAT ", " |
| "written = %" G_GSSIZE_FORMAT, ret, written); |
| break; |
| } |
| } |
| |
| gst_buffer_unmap (outbuf, &out_map); |
| |
| if (i > 0) |
| gst_buffer_set_size (outbuf, i * frame_len); |
| else |
| gst_buffer_replace (&outbuf, NULL); |
| } |
| |
| done: |
| |
| gst_buffer_unmap (buffer, &in_map); |
| |
| return gst_audio_encoder_finish_frame (audio_enc, outbuf, |
| i * (samples_per_frame / enc->channels)); |
| |
| /* ERRORS */ |
| no_buffer: |
| { |
| GST_ERROR_OBJECT (enc, "could not allocate output buffer"); |
| goto done; |
| } |
| map_failed: |
| { |
| GST_ERROR_OBJECT (enc, "could not map input buffer"); |
| goto done; |
| } |
| } |
| |
| static gboolean |
| gst_sbc_enc_start (GstAudioEncoder * audio_enc) |
| { |
| GstSbcEnc *enc = GST_SBC_ENC (audio_enc); |
| |
| GST_INFO_OBJECT (enc, "Setup subband codec"); |
| sbc_init (&enc->sbc, 0); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_sbc_enc_stop (GstAudioEncoder * audio_enc) |
| { |
| GstSbcEnc *enc = GST_SBC_ENC (audio_enc); |
| |
| GST_INFO_OBJECT (enc, "Finish subband codec"); |
| sbc_finish (&enc->sbc); |
| |
| enc->subbands = 0; |
| enc->blocks = 0; |
| enc->rate = 0; |
| enc->channels = 0; |
| enc->bitpool = 0; |
| |
| return TRUE; |
| } |
| |
| static void |
| gst_sbc_enc_class_init (GstSbcEncClass * klass) |
| { |
| GstAudioEncoderClass *encoder_class = GST_AUDIO_ENCODER_CLASS (klass); |
| GstElementClass *element_class = GST_ELEMENT_CLASS (klass); |
| |
| encoder_class->start = GST_DEBUG_FUNCPTR (gst_sbc_enc_start); |
| encoder_class->stop = GST_DEBUG_FUNCPTR (gst_sbc_enc_stop); |
| encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_sbc_enc_set_format); |
| encoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_sbc_enc_handle_frame); |
| |
| gst_element_class_add_pad_template (element_class, |
| gst_static_pad_template_get (&sbc_enc_sink_factory)); |
| |
| gst_element_class_add_pad_template (element_class, |
| gst_static_pad_template_get (&sbc_enc_src_factory)); |
| |
| gst_element_class_set_static_metadata (element_class, |
| "Bluetooth SBC audio encoder", "Codec/Encoder/Audio", |
| "Encode an SBC audio stream", "Marcel Holtmann <marcel@holtmann.org>"); |
| |
| GST_DEBUG_CATEGORY_INIT (sbc_enc_debug, "sbcenc", 0, "SBC encoding element"); |
| } |
| |
| static void |
| gst_sbc_enc_init (GstSbcEnc * self) |
| { |
| self->subbands = 0; |
| self->blocks = 0; |
| self->rate = 0; |
| self->channels = 0; |
| self->bitpool = 0; |
| } |