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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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baseaudioencoder: port to new GstAudioInfo API
This commit is contained in:
Tim-Philipp Müller
parent
c89b49bfaf
commit
7f0c7e5f82
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@ -182,7 +182,7 @@ enum
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typedef struct _GstBaseAudioEncoderContext
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{
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/* input */
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GstAudioFormatInfo info;
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GstAudioInfo info;
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/* output */
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gint frame_samples;
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@ -410,7 +410,7 @@ gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
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enc->priv->active = FALSE;
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enc->priv->samples_in = 0;
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enc->priv->bytes_out = 0;
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gst_base_audio_format_info_clear (&enc->priv->ctx.info);
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gst_audio_info_clear (&enc->priv->ctx.info);
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memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
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}
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@ -935,14 +935,32 @@ wrong_buffer:
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}
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}
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static gboolean
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audio_info_is_equal (GstAudioInfo * from, GstAudioInfo * to)
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{
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if (from == to)
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return TRUE;
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if (GST_AUDIO_INFO_FORMAT (from) != GST_AUDIO_INFO_FORMAT (to))
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return FALSE;
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if (GST_AUDIO_INFO_RATE (from) != GST_AUDIO_INFO_RATE (to))
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return FALSE;
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if (GST_AUDIO_INFO_CHANNELS (from) != GST_AUDIO_INFO_CHANNELS (to))
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return FALSE;
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if (GST_AUDIO_INFO_CHANNELS (from) > 64)
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return TRUE;
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return memcmp (from->position, to->position,
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GST_AUDIO_INFO_CHANNELS (from) * sizeof (to->position[0]));
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}
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static gboolean
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gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstBaseAudioEncoder *enc;
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GstBaseAudioEncoderClass *klass;
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GstBaseAudioEncoderContext *ctx;
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GstAudioFormatInfo *state, *ostate;
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GstAudioInfo *state, *old_state;
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gboolean res = TRUE, changed = FALSE;
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guint old_rate;
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enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
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klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
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@ -959,18 +977,19 @@ gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
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goto refuse_caps;
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/* adjust ts tracking to new sample rate */
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if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && state->rate) {
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old_rate = GST_AUDIO_INFO_RATE (state);
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if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
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enc->priv->base_ts +=
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GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, state->rate);
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GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
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enc->priv->samples = 0;
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}
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ostate = gst_base_audio_format_info_copy (state);
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if (!gst_base_audio_parse_caps (caps, state))
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old_state = gst_audio_info_copy (state);
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if (!gst_audio_info_from_caps (state, caps))
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goto refuse_caps;
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changed = gst_base_audio_compare_format_info (state, ostate);
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gst_base_audio_format_info_free (ostate);
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changed = audio_info_is_equal (state, old_state);
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gst_audio_info_free (old_state);
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if (changed) {
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GstClockTime old_min_latency;
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@ -1225,8 +1244,8 @@ gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
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gint64 src_val, dest_val;
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gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
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if (!(res = gst_base_audio_raw_audio_convert (&enc->priv->ctx.info,
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src_fmt, src_val, &dest_fmt, &dest_val)))
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if (!(res = gst_audio_info_convert (&enc->priv->ctx.info,
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src_fmt, src_val, dest_fmt, &dest_val)))
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goto error;
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gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
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break;
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@ -1255,6 +1274,77 @@ gst_base_audio_encoder_get_query_types (GstPad * pad)
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return gst_base_audio_encoder_src_query_types;
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}
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/*
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* gst_base_audio_encoded_audio_convert:
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* @fmt: audio format of the encoded audio
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* @bytes: number of encoded bytes
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* @samples: number of encoded samples
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* @src_format: source format
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* @src_value: source value
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* @dest_format: destination format
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* @dest_value: destination format
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*
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* Helper function to convert @src_value in @src_format to @dest_value in
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* @dest_format for encoded audio data. Conversion is possible between
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* BYTE and TIME format by using estimated bitrate based on
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* @samples and @bytes (and @fmt).
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*/
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/* FIXME: make gst_base_audio_encoded_audio_convert() public? */
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static gboolean
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gst_base_audio_encoded_audio_convert (GstAudioInfo * fmt,
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gint64 bytes, gint64 samples, GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
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{
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gboolean res = FALSE;
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g_return_val_if_fail (dest_format != NULL, FALSE);
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g_return_val_if_fail (dest_value != NULL, FALSE);
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if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
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src_value == -1)) {
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if (dest_value)
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*dest_value = src_value;
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return TRUE;
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}
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if (samples == 0 || bytes == 0 || fmt->rate == 0) {
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GST_DEBUG ("not enough metadata yet to convert");
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goto exit;
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}
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bytes *= fmt->rate;
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switch (src_format) {
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_TIME:
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*dest_value = gst_util_uint64_scale (src_value,
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GST_SECOND * samples, bytes);
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res = TRUE;
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break;
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default:
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res = FALSE;
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}
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break;
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = gst_util_uint64_scale (src_value, bytes,
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samples * GST_SECOND);
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res = TRUE;
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break;
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default:
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res = FALSE;
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}
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break;
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default:
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res = FALSE;
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}
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exit:
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return res;
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}
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/* FIXME ? are any of these queries (other than latency) an encoder's business
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* also, the conversion stuff might seem to make sense, but seems to not mind
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* segment stuff etc at all
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@ -1481,13 +1571,13 @@ gst_base_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
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}
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/**
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* gst_base_audio_encoder_get_info:
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* gst_base_audio_encoder_get_audio_info:
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* @enc: a #GstBaseAudioEncoder
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*
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* Returns: a #AudioFormatInfo describing input audio format
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* Returns: a #GstAudioInfo describing the input audio format
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*/
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GstAudioFormatInfo *
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gst_base_audio_encoder_get_info (GstBaseAudioEncoder * enc)
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GstAudioInfo *
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gst_base_audio_encoder_get_audio_info (GstBaseAudioEncoder * enc)
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{
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g_return_val_if_fail (GST_IS_BASE_AUDIO_ENCODER (enc), NULL);
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@ -28,7 +28,7 @@
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#endif
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#include <gst/gst.h>
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#include <gst/audio/gstbaseaudioutils.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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@ -158,7 +158,7 @@ struct _GstBaseAudioEncoderClass {
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gboolean (*stop) (GstBaseAudioEncoder *enc);
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gboolean (*set_format) (GstBaseAudioEncoder *enc,
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GstAudioFormatInfo *info);
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GstAudioInfo *info);
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GstFlowReturn (*handle_frame) (GstBaseAudioEncoder *enc,
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GstBuffer *buffer);
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@ -187,7 +187,7 @@ GstCaps * gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc,
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/* context parameters */
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GstAudioFormatInfo * gst_base_audio_encoder_get_info (GstBaseAudioEncoder * enc);
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GstAudioInfo * gst_base_audio_encoder_get_audio_info (GstBaseAudioEncoder * enc);
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gint gst_base_audio_encoder_get_frame_samples (GstBaseAudioEncoder * enc);
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void gst_base_audio_encoder_set_frame_samples (GstBaseAudioEncoder * enc,
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